Abstract: An electronic apparatus is provided. The electronic apparatus includes a microphone and a processor configured to, based on a second sound including a first sound of an audio signal being received through the microphone, obtain a noise about the electronic apparatus based on the audio signal and the second sound, obtain an enduring component of the noise, and based on the enduring component being equal to or greater than a threshold value, process the audio signal.

Abstract: A noise reduction device includes a processor that converts a noise signal collected by a microphone disposed in a control space into a noise signal in a frequency domain, a storage that stores the converted noise signal in the frequency domain as a reference signal, and a signal generator that generates a noise reduction signal for reducing the noise signal collected by the microphone at a control position of the control space. The processor determines whether or not the noise signal is non-stationary noise based on a frequency characteristic of the converted noise signal in the frequency domain and a frequency characteristic of the reference signal. When it is determined that the noise signal is the non-stationary noise, the processor controls the signal generator so as to cancel generation of the noise reduction signal.

Abstract: A signal processor and a method for processing an input signal are presented. The signal processor is adapted to clip an oversampled input signal without introducing noise in the frequency band of interest. For instance, the signal processor may be used for clipping an acoustic signal. The signal processor includes a summer coupled to a limiter and to a feedback circuit. The summer is adapted to sum the input signal with at least one feedback signal to provide an adjusted signal. The limiter is adapted to compare the adjusted signal with a first threshold value and a second threshold value to provide a limited signal. The feedback circuit is adapted to calculate a difference between the limited signal and the adjusted signal, and to generate at least one feedback signal based on the difference.

Abstract: A lidar system includes a transmitter that encodes successive transmit pulses with different pulse characteristics and a receiver that detects the pulse characteristics of each received (scattered or reflected) pulse and that distinguishes between the received pulses based on the detected pulse characteristics. The lidar system thus resolves range ambiguities by encoding pulses of scan positions in the same or different scan periods to have different pulse characteristics, such as different pulse widths or different pulse envelope shapes. The receiver includes a pulse decoder configured to detect the relevant pulse characteristics of the received pulse and a resolver that determines if the pulse characteristics of the received pulse matches the pulse characteristics of the current scan position or that of a previous scan position.

Abstract: A system used in a loudspeaker-room-microphone environment includes a microphone signal partitioner that divides a signal from a microphone into one or more divided portions. A reverberation energy estimator estimates reverberation energy in some of the divided portions of the microphone signal based on a loudspeaker signal. The estimated reverberation energy is processed to generate a dereverberated output signal.

Abstract: A system may include a set of 4N packet processors and a switching fabric to interconnect the set of 4N packet processors. The switching fabric may include the following switching elements having a size of at least 3N×3N: a first switching element, a second switching element, a third switching element, and a fourth switching element. The first switching element may be directly connected to the second switching element and the third switching element, and may be indirectly connected to the fourth switching element. The second switching element may be directly connected to the fourth switching element, and may be indirectly connected to the third switching element. The third switching element may be directly connected to the fourth switching element.

Abstract: Beacon devices may transmit beacon messages to alert an application on a mobile phone of the beacon device's proximity. An encrypted authorization may be used to prevent malicious attacks. Unfortunately, some operating systems, e.g., the iOS™ operating system, strip data fields in service Universally Unique Identifiers (UUIDs) from a received beacon message before the message is passed to the application. Various embodiments provide a protocol wherein the beacon device successively transmits a first beacon message (e.g., an iBeacon™ message) and a second Bluetooth® Low Energy broadcast message. The first beacon message alerts the application of the beacon device's proximity and identity while the second message contains the encrypted authorization and possibly any additional information relevant to the application.

Abstract: Systems and methods are described for storing and reusing previously generated/calculated acoustic environment data. By reusing acoustic environment data, the systems and methods described herein may avoid the increased overhead in generating/calculating acoustic environment data for a location when this data has already been generated and is likely accurate. In particular, the time and complexity involved in determining reverberation/echo levels, noise levels, and noise types may be avoided when this information is available in storage. This previously stored acoustic environment data may not be limited to data generated/calculated by the same audio device. Instead, in some embodiments an audio device may access a centralized repository to leverage acoustic environment data generated/calculated by other audio devices.

Abstract: In one embodiment, an echo canceller configured to cancel echo in a wideband voice conference is provided. A double-talk condition may be when a plurality of users are speaking substantially simultaneously. When a double-talk condition is detected in the wideband conference, a high-frequency process is enabled and used to process signals in the high band to reduce echo. Accordingly, echo in the high band may not be produced by end devices being used by the users' speaking. Also, the users speaking have the echo cancelled in the low band and substantial echo does not result. This results in the users speaking experiencing the conference in the narrowband. The other users that are not speaking, however, continue to receive wideband signals. The users not speaking also continue to have echo cancellation performed for the high band and low band because these users are not speaking and thus attenuation of their voices is not a consideration.

Abstract: A system and method for compensating audio signal components is disclosed. The method includes detecting, by a microphone, a sound signal. The sound signal comprises audio signal components resulting from reproducing an audio signal of an audio source and speech signal components corresponding to a speech signal from a person. The sound signal is filtered to whiten the sound signal. The audio signal components in the whitened sound signal are then compensated. The whitening of the compensated sound signal is removed. The filtering of the audio signal is performed using at least two filters in an alternating way, each filter using time-dependent filter coefficients.

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: A transceiver system is disclosed. The transceiver system comprises a first transceiver physical layer circuit (PHY) having a first plurality of channels and a second transceiver PHY disposed adjacent the first transceiver PHY and having a second plurality of channels. Filter circuitry is coupled between at least one of the plurality of first channels and at least one of the plurality of second channels.

Abstract: A transceiver mitigates signal leakage into a receive path from a transmit path. A subtraction circuit determines a difference between a receive signal and a compensation signal to produce a compensated receive signal prior to demodulation by a demodulator. An equalizer both amplitude adjusts and phase adjusts orthogonal baseband transmit signals based on the difference from the subtraction circuit to produce the compensation signal. A digital tuning circuit determines at least one amplitude adjust coefficient to be used by the equalizer. The equalizer can have a polarity switch or a variable attenuator or a variable delay.

Abstract: A method for identifying supported transmission channel configurations in a mobile communications apparatus, the method comprising: calculating the required power value for each transmission channel configuration; and performing a sequential comparison of: an indication of an available measured power value, and an indication of the required power for each transmission channel configuration; and identifying a particular transmission channel configuration as supported if the required power for the particular transmission channel configuration is less than the available power value; wherein: the sequential comparison is performed in order of magnitude of value of the indication of the required power; and the method further comprising stopping said comparison when a supported transmission channel configuration is identified.

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: Embodiments of a multi-channel audio communication device and methods for reducing echoes are generally described herein. Other embodiments may be described and claimed. In some embodiments, echo-reduction circuitry may insert training signals into digital audio signals at or below a noise floor and in a non-audible portion of the frequency spectrum based on a spectral mask. Training signals may be generated from spectrum estimates and training sequences. An adaptive filter may generate echo-cancellation signals using filter coefficients generated fawn the training sequences and return-path signals. The echo cancellation signals may remove echo signals from the return-path signals.

Abstract: Embodiments of an audio communication device and methods for reducing echoes are generally described herein. Other embodiments may be described and claimed. In some embodiments, echo-reduction circuitry may insert a training signal into digital audio signals at or below a noise floor and in a non-audible portion of the frequency spectrum based on a spectral mask. The training signal may be generated from a spectrum estimate and a training sequence. An adaptive filter may generate an echo-cancellation signal using filter coefficients generated from the training sequence and return-path signals. The echo-cancellation signal may remove echo signals from the return-path signals.

Abstract: A system for suppressing transmission channel noise in a signal transmission/reception device—particularly a time division duplex (TDD) wireless communications device—is provided. The system provides one or more shunt elements, instantiated at some point along the device's signal transmission channel. One or more attenuation elements are also instantiated at some point along the signal transmission channel, as are one or more disabling elements. A trigger signal indicates when the device is shifting from signal transmission operation (or mode) to signal reception operation. Responsive to assertion of the trigger signal, the shunt, attenuation and disabling elements are activated.

Abstract: A system for suppressing transmission channel noise in a signal transmission/reception device—particularly a time division duplex (TDD) wireless communications device—is provided. The system provides one or more shunt elements, instantiated at some point along the device's signal transmission channel. One or more attenuation elements are also instantiated at some point along the signal transmission channel, as are one or more disabling elements. A trigger signal indicates when the device is shifting from signal transmission operation (or mode) to signal reception operation. Responsive to assertion of the trigger signal, the shunt, attenuation and disabling elements are activated.

Abstract: The present invention relates to a method for attenuating near-end crosstalk between a remote received signal and a locally transmitted signal in a bidirectional signal transmission wherein said method comprises the step of producing an analog automatic feedback-driven correction loop in order to provide a no-correlation between said transmitted and received signals and to a hybrid circuit for enabling said method.

Abstract: A system and method for designing a full duplex communications transceiver to enable a reduction in isolation between a transmit signal and a receive signal at a diplexer is disclosed. An RFE transmit filter attenuation level can be determined to be sufficient to attenuate a noise power of the transmitter below a thermal noise floor of the power amplifier. A minimum diplexer transmit filter attenuation level can be calculated to attenuate noise power output from the power amplifier in the receive band to a level below a noise floor of the receiver.

Abstract: A transmission method includes transmitting a signal using an orthogonal frequency division multiplexing, receiving the transmitted signal at a plurality of reception devices, and controlling a transmission power for first signals transmitted to the plurality of reception devices in an area formed by the transmitter to be greater than a transmission power for second signals transmitted to a specific device in the area, wherein said first signals and said second signals are time division multiplexed.

Abstract: A system processes signals exchanged between a near end terminal and a far end terminal over a communication path. The system selects the processing characteristics of a near end terminal based on characteristics of the communication path. The communication path may include the near and the far end terminal and their respective codecs, and the networks. The system may select processing characteristics of the near end terminal based on characteristics of the communication path. Selecting the processing characteristics may include selecting a predetermined operation. The system may adapt a selected operation based on a real time analysis of the communication path.

Abstract: A system and method for designing a full duplex communications transceiver to enable a reduction in isolation between a transmit signal and a receive signal at a diplexer is disclosed. An RFE transmit filter attenuation level can be determined to be sufficient to attenuate a noise power of the transmitter below a thermal noise floor of the power amplifier. A minimum diplexer transmit filter attenuation level can be calculated to attenuate noise power output from the power amplifier in the receive band to a level below a noise floor of the receiver.

Abstract: The present invention is directed to implementing theoretical foundations of frequency domain methods aiming to locally decrease the impact of aggregate side lobes of a few discrete frequency tones, in the vicinity of other “victim” frequencies. The family of frequency methods referred to makes use of one or more “canceller tones.” The “canceller tones” loads are optimized to minimize the aggregate power in the neighborhood of the victim frequencies.

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: A method is provided for introducing analog loss to an information-bearing signal being transmitted to or from end-user terminal equipment over a communication system such as a cable network The method begins by establishing a communication path to the terminal equipment over the communication system and then introducing a level of loss into the communication path to reduce echo. The level of loss that is introduced is based on the information content of the information-bearing signal.

Abstract: This invention provides a technique to reduce echoes not adequately attenuated by echo cancellers during initial periods of voice communication. Signals transmitted across communication networks are often reflected back to the caller resulting in echo. Echo cancellers are employed in the communication system to cancel this effect in order to maintain a high quality transmission. However, echo cancellers require time to detect, adapt, and effectively remove echo, often resulting in echo during the initial moments of the call and thereby degrading the quality of service. By installing an attenuation device into the communication system, all signals that pass through it are reduced to a preset value for a set period of time. This reduces any echo below a detectable threshold. After the period of time expires, the attenuation device allows the signal to pass unaffected, by which time the echo cancellers have adaptively adjusted to any echoes in the system.

Abstract: System for testing attenuators by a flatness and standing wave ratio tests which includes a vector network analyzer (VNA) adapted to be coupled to a device under test (DUT) and which provides an input stimulus signal for the DUT and, when certain conditions are satisfied, receives an output signal from the DUT, and a calibration receiver adapted to be coupled to the DUT via a down-converter. When certain conditions are satisfied, the output signal from the DUT is sent to the calibration receiver (through the down-converter to be possibly modified thereby depending on the testing frequency). A signal generator provides a local oscillator (LO) signal for the down-converter. A control unit is connected to the instruments and embodies software which analyzes the testing conditions, i.e., the attenuator value being tested, and selects whether the network analyzer or the calibration receiver will measure the output signal from the DUT.

Abstract: An echo cancellation system for detecting and canceling echo from a network speech path, comprising a pool of echo locators for detecting pure echo delay and dispersion of time, a pool of echo cancellers, a pool of line watchdogs for detecting presence or absence of echo in the network speech path, and a control system for (i) locating and applying an echo locator from the pool of echo locators to the network speech path for detecting the pure echo delay and dispersion of time of any echo in the network speech path, and thereafter returning the echo locator to the pool of echo locators; (ii) locating and applying an echo canceller from the pool of echo cancellers to the network speech path for canceling the echo using the pure delay and dispersion time, and thereafter returning the echo canceller to the pool of echo cancellers; and (iii) locating and applying a line watchdog from the pool of line watchdogs to the network speech path for detecting said presence or absence of echo, and in the event the presence

Abstract: An echo cancelling system for cancelling echoes in a communication path includes an echo locator and an echo canceller. The echo locator is coupled to the communication path and locates the positions of echo signals received from the communication path. The echo canceller is also coupled to the communication path and cancels echo signals received therefrom. During echo cancellation, the echo canceller becomes active in regions corresponding to the positions of echo signals in response to the echo locator.

Abstract: A method and apparatus for reducing echo feedback in a wireless communication system (10) is accomplished when a receiving communication unit (24) senses a feedback signal, or echo, via an ancillary communication path. The receiving communication unit is a targeted recipient of an original audio signal generated by a transmitting communication unit (22), where the original audio signal (42) is conveyed to the receiving communication unit. Upon detecting the feedback signal and determining that it exceeds a feedback threshold, the receiving communication unit attenuates an audible output of the original signal to reduce echo to the transmitting communication unit. In addition, the receiving communication unit, and/or the transmitting communication unit include echo canceller to further minimize the echo within the digital communication system.

Abstract: An acoustic echo control system and a double talk control method thereof that eliminate echo signals which are generated in a hand-free communication system. In order to discriminate a double talk generation and a change of an echo path, the acoustic echo control system accurately detects a double talk section and an echo path change section using a near end signal from a near end talker and a far end signal from a far end talker and in the double talk section an adaptive echo remover of the system suspends a filter coefficient adaption and estimates an acoustic echo path which is supplied to a microphone over a speaker by performing the filter coefficient adaption in the echo path change section, thereby improving communication quality and discriminating the double talk and the change of the echo path without time delay.

Abstract: The invention relates to data transmission in telecommunication systems and particularly in radio systems. The invention employs “payload numbering” instead of or in addition to conventional frame numbering. Data (61) is split into fixed-length data blocks or payload units (62). The size of a block is preferably equal to or smaller than the shortest information field in frames (63) of the protocol(s) used. Each protocol frame carries one or more payload units. In an optimum situation the length of the information field in a protocol frame equals n * the length of the payload unit, where n is an integer. Instead of frame numbering (in some special cases possibly in addition to it) the protocol frame carries payload numbers both for indicating the payload units (data blocks) conveyed in the protocol frame and for acknowledging the received blocks.

Abstract: A system and method provide ASR talkoff suppression in a prompt-and collect voice transaction system in which a prompt signal is sent and an input signal is received. A comparison is made of a characteristic of the prompt signal and the input signal. Processing of the input signal to reduce the likelihood of an ASR response thereto is performed if the result of the comparison satisfies a predetermined criterion.

Abstract: A method and apparatus for reducing echo feedback in a wireless communication system (10) is accomplished when a receiving communication unit (24) senses a feedback signal, or echo, via an ancillary communication path. The receiving communication unit is a targeted recipient of an original audio signal generated by a transmitting communication unit (22), where the original audio signal (42) is conveyed to the receiving communication unit. Upon detecting the feedback signal and determining that it exceeds a feedback threshold, the receiving communication unit attenuates an audible output of the original signal to reduce echo to the transmitting communication unit. In addition, the receiving communication unit, and/or the transmitting communication unit include echo canceller to further minimize the echo within the digital communication system.

Abstract: A system and method for the efficient transmission of information in a code division multiple access (CDMA) wireless telecommunication system. The rate of reliable transmission is increased by implementing an orthogonal frequency-division multiplexing (OFDM) scheme in, for example, a direct-spread CDMA network, resulting in a multi-carrier CDMA (MC-CDMA) system. Information (such as voice and data), is encoded, divided, and spread across the frequency domain, rather than in the time domain as in traditional CDMA; the allowable transmission bandwidth is divided into a number of carriers. Using this scheme, a number of loading parameters such as code rate, data rate, and the number of streams into which the encoded data is divided may be varied to increase the performance of the system. Application of the variable loading parameter may be a function of channel quality, such as the presence of noise or the channel fading state.

Abstract: An interference canceller includes despread processing parts, a combiner combining interference replica generation signals, a decision part that decides an output signal, spread processing parts coupled to the despread processing parts and the decision part, an attenuation coefficient generator generating an attenuation coefficient dependent on a reliability of the interference replica generation signals, and a multiplier multiplying the output signal of the decision part by the attenuation coefficient.

Abstract: The present invention, generally speaking, provides a digitally-tunable, echo-cancelling analog front end (AFE) for wireline digital communications. The analog front end is especially useful in a High-bit-rate Digital Subscriber Line (HDSL) or HDSL2 environment. An analog echo simulation path is provided capable of simulating echo from a wide variety of echo paths. Digitally controlled attenuators are provided in the transmission path and in the analog echo simulation path. Also provided is a digital-tunable equalizer stage. The equalizer stage is tuned to match the characteristics of the receive path. The same arrangement may be adapted for various DSL technologies, i.e., xDSL. There results an analog front end that is well-adapted to high-speed wireline communications.

Abstract: In a data communications system, a first node transmits an output digital signal to a second node through a transmission line and receives an input digital signal from the second node through a reception line, a signal processing amplification block of the data communications system compensates an attenuation in the input digital signal and prevents a crosstalk between the transmission line and the reception line, and a regulating block for preventing a crosstalk receives a branch signal and generates a control signal based on a capacity of the branch signal.

Abstract: A startup protocol is provided for use in a communications system having a plurality of transceivers, one transceiver acting as a master and another transceiver acting as a slave, each transceiver having a noise reduction system, a timing recovery system and at least one equalizer. The operation of the startup protocol is partitioned into three stages. During the first stage the timing recovery system and the equalizer of the slave are trained and the noise reduction system of the master is trained. During the second stage the timing recovery system of the master is trained in both frequency and phase, the equalizer of the master is trained and the noise reduction system of the slave is trained. During the third stage the noise reduction system of the master is retrained, the timing recovery system of the master is retrained in phase and the timing recovery system of the slave is retrained in both frequency and phase.

Abstract: In a case where transmission and reception timing of a base station has not been synchronized to that of another base station in the vicinity, or in a case where it is necessary to establish such synchronization, a synchronous controller outputs a reception-mode changeover signal to change the prevailing mode from an ordinary reception mode to a synchronous control reception mode. As a result, attenuating means attenuates a strong input signal from the other base station and enters the attenuated signal into a demodulator via a front end. The demodulator proceeds to demodulate the received data. The synchronous controller detects synchronization information (a synchronizing word) from the received data to synchronize the transmission and reception timings of the two base stations and then returns the reception mode to the ordinary reception mode.

Abstract: This invention deals with an adaptive echo cancelling system for communication networks. The system includes an adaptive filter for acoustic echo cancellation that uses a combination of echo cancelling and gain control techniques. The main advantages of the combination of these approaches are good communication quality during double talk and easier control of the output echo level.

Abstract: Apparatus and method for the prevention of echo signals in terminals connected to a network carrying telephony traffic. The echo prevention occurs prior to the introduction of signal delay by a network. The method and apparatus function to control echo contained in audio signals transmitted between the nodes of a network, wherein the audio signals experience delay in transit from node to node and wherein the echo controlling method is distributed across the nodes.

Abstract: A telecommunication signal processor includes a noise compensation system for preventing undesirable spectral modulation of the noise component of a telecommunication signal subjected to a process wherein subband portions of the signal are variably attenuated. A noise component is injected into each attenuated error signal in order to compensate for attenuation of background noise incidental to the variable attenuation of the subband signals. The amplitude of the noise components are each determined independently on the basis of an estimated noise content of the original subband signals and in accordance with the applied attenuation within each subband. The attenuated and noise-compensated subband signals are combined by a synthesis filter to provide a send-output signal.

Abstract: A variable gain echo suppressor is provided that utilizes two variable attenuators (410) and (412) that are disposed in the receive and transmit paths, respectively, of a communication system. An echo suppressor (414) receives the input from the far end and transmits the output from the near end and controls the two attenuators to set a variable gain therein. The echo suppressor utilizes a normalized power level which is normalized with respect to the background noise at the respective end and then takes the ratio of this normalized power and the normalized power at the opposite end to define an associated power ratio. This power ratio value is utilized to determine the attenuation for the associated end. When the power ratio is high, this indicates that a stronger signal is being received from that end and a small amount of attenuation is associated in the transmit path from that end. When they aye equal, this will result in both systems having approximately the same attenuation, a mid-level attenuation.

Abstract: A control device for a hands-free telephone set automatically controls microphone and amplifier gains so that a feedback loop has less than unity gain to avoid circuit instability and resultant self-oscillation. An emission channel includes a microphone, a signal compressor and a controllable attenuator. A reception channel includes a signal compressor, an adjustable attenuator and a loudspeaker. The combination of the emission and reception channels form an amplification loop whereby the output of the reception channel is acoustically coupled to the input of the emission channel while the output of the emission channel is coupled to the input of the reception channel through a common telephone line. To avoid circuit oscillation, a circuit initially sets the gain of the loop to a predetermined value slightly less than unity (0 db) and subsequently maintains the loop gain constant by maintaining the sum of the compressor and attenuator gains at a fixed value.