Abstract: Exemplary embodiments include methods and systems for detecting an incoming telephone signal, processing the incoming telephone signal, transmitting the processed signal to a remote receiver and alerting a remote user of the incoming signal.

Abstract: A soft reset method for a DSL apparatus. The method includes: receiving a soft reset command; controlling, according to the soft command, a service module of the DSL apparatus to perform a soft reset; and simultaneously controlling, according to a preset service module type, a non-service module of the DSL apparatus to maintain a working status. According to at least some embodiments of the invention, during the soft reset process of the DSL apparatus, it is unnecessary to perform reset on all service modules and non-service modules, so that an initialization operation load of the system during the soft reset process is reduced. Moreover, because it is unnecessary to perform soft reset on non-service modules, time that is consumed during the soft reset process of the DSL apparatus is reduced, and the user experience is correspondingly enhanced.

Abstract: An communication circuit of a communication device includes an interface unit, a direct current (DC) control circuit, a voice output circuit includes a voice input terminal, and a voice receiving circuit includes a voice output terminal. The interface unit outputs human speech and receives human speech from another communication device. The DC control circuit provides a constant current for the optocoupler circuit. When the outgoing human speech is coupled to the DC control circuit by a first optocoupler and sent to another communication device and incoming human speech is transmitted from the other communication device via the DC control circuit and a second optocoupler, echoes and extraneous noise in the speech is reduced.

Abstract: An apparatus for removing an echo(es) from audio content may include a processor and memory storing executable computer code causing the apparatus to at least perform operations including receiving combined audio content including voice data associated with speech of users in a call and information including audio data provided to the users. The computer program code may further cause the apparatus to remove a first echo of a first item of voice data associated with a user(s), from the combined audio content, based in part on a prior detection of the first item of voice data. The computer program code may further cause the apparatus to remove a second echo of the audio data, from the combined audio content, based in part on a previous detection of the audio data or a previous detection of data corresponding to the audio data. Corresponding methods and computer program products are also provided.

Abstract: The present invention provides for echo cancellation circuitry and variable rate encoding circuitry to cooperate with one another to effectively provide comfort noise in an effective and efficient manner. The echo cancellation circuitry will use far-end signals to generate estimated echo signals, which correspond to the actual echo signals appearing in near-end signals. The estimated echo signals are essentially subtracted from the near-end signals in an effort to remove the actual echo signals from the near-end signals. The echo cancellation circuitry will monitor any residual echo signals in the resulting processed near-end signals and provide residual echo control signals that are indicative of whether the residual echo signals should be replaced with comfort noise. The residual echo control signals are used at least in part by the variable rate encoding circuitry to determine the encoding rate to use for encoding different portions of the processed near-end signals.

Abstract: A method and a corresponding device for reducing enhanced noise generated as a result of applying vectoring to signals transmitted and subsequently received over a plurality of copper telephone loops involve receiving a set of intermediate values, which correspond to estimates of the transmitted signals. The intermediate values are generated by applying a vectoring procedure to initial samples of the received signals. The intermediate values are demodulated. At least one demodulated value is selected as a final estimate of a corresponding transmitted signal. The initial samples are then refined by removing each initial sample that has a corresponding selected demodulated value. The vectoring procedure, the demodulating, the selecting, and the refining are each repeated using the refined samples in place of the initial samples to generate new intermediate values that have reduced noise.

Abstract: An embodiment of a method for routing an incoming call includes receiving an incoming call addressed to a single phone number associated with a plurality of users at a network routing device. The method further includes determining, by the network routing device, an incoming caller identifier associated with the incoming call and determining, by the network routing device whether the incoming caller identifier matches at least one caller identifier in a caller list associated with at least one particular user of the plurality of users. If the caller identifier matches a caller identifier in a caller list associated with the particular user, the incoming call is routed, by the network routing device, to at least one telephone associated with the particular user.

Abstract: An Ethernet extension device is provided for metro or last mile Ethernet service via twisted pairs as opposed to fiber optics. The Ethernet extension device is implemented as a plug-in extension for existing infrastructure (e.g., in a standard electrical wall box or Type-200™ Mechanics card) that employs lighting and power cross protection required by the telephone companies for Ethernet connectivity to the telephone network (e.g., for connection between a user's building and a telephone company building over existing outdoor telephone cables).

Abstract: In one embodiment, an acoustic echo control (AEC) module receives an outgoing signal and an incoming signal, which, at various times, contains acoustic echo corresponding to the outgoing signal. The AEC module has a delay estimation block that estimates, in the time domain, the echo delay using an adaptive filtering technique. This delay estimation is used to align samples of the incoming signal having acoustic echo with the corresponding samples of the outgoing signal from which the acoustic echo originated. The AEC module determines whether or not samples of the incoming signal contain acoustic echo based on the aligned outgoing signal, and the determinations are applied to a hangover counter. The AEC module then suppresses acoustic echo in the incoming signal and adds comfort noise to the incoming signal. The amount of echo suppression performed is gradually increased or decreased based on comparisons of the counter to a hangover threshold.

Abstract: A telecommunication system including a fall duplex speakerphone, comprising a first microphone to generate a coupled signal including uplink information and non-linear distortion sensed by the first microphone in a speaker phone mode, a second microphone to generate a reference signal including downlink information and the non-linear distortion sensed by the second microphone in the speaker phone mode, and an acoustic echo canceller (AEC) to receive the coupled signal from the first microphone, to receive the reference signal from the second microphone, and to cancel out the non-linear distortion included in the coupled signal based on the non-linear distortion included in the reference signal.

Abstract: Techniques for recording and replay of a live conference while still attending the live conference are described. A conferencing system includes a user interface generator, a live conference processing module, and a replay processing module. The user interface generator is configured to generate a user interface that includes a replay control panel and one or more output panels. The live conference processing module is configured to extract information included in received conferencing data that is associated with one or more conferencing modalities, and to display the information in the one or more output panels in a live manner (e.g., as a live conference). The replay processing module is configured to enable information associated with the one or more conferencing modalities corresponding to a time of the conference session prior to live to be presented at a desired rate, possibly different from the real-time rate, if a replay mode is selected in the replay control panel.

Abstract: The invention provides a method for determining a set of filter coefficients for an acoustic echo compensator in a beamformer arrangement. The acoustic echo compensator compensates for echoes within the beamformed signal. A plurality of sets of filter coefficients for the acoustic echo compensator is provided. Each set of filter coefficients corresponds to one of a predetermined number of steering directions of the beamformer arrangement. The predetermined number of steering directions is equal to or greater than the number of microphones in the microphone array. For a current steering direction, a current set of filter coefficients for the acoustic echo compensator is determined based on the provided sets of filter coefficients.

Abstract: An apparatus comprising a processor configured to determine a power management (PM) state from a plurality of PM states for a digital subscriber line (DSL) transceiver, wherein the plurality of PM states comprise a idle state, a full on state, and a low power state, wherein the full on state consumes more power than the low power state, wherein the low power state consumes more power than the idle state, wherein the low power state comprises at least one PM profile, and wherein the DSL transceiver supports a very high speed digital subscriber line 2 (VDSL2) data transmission protocol, and instruct the DSL transceiver to use the PM state.

Abstract: A CODEC circuit for POTS system comprises a digital circuit for outputting a digital control signal having a first bit width, the digital control signal being indicative of a voltage to be applied to a POTS subscriber line pair. Further, the CODEC comprises a noise shaper coupled to an output of the digital circuit for generating a noise-shaped control signal and a digital-to-analog converter coupled to an output of the noise shaper, the input of the digital-to-analog converter having a second bit width being larger than 1 and smaller than the first bit width.

Abstract: Selective audio combination for a conference. The conference may be initiated between a plurality of participants at respective participant locations. The conference may be performed using a plurality of conferencing endpoints at each of the participant locations. Audio may be received from each of the plurality of conferencing endpoints. Audio level information may also be received from each of the plurality of conferencing endpoints. The audio may be combined from a plural subset of the plurality of conferencing endpoints to produce conference audio. The plural subset is less than all of the plurality of conferencing endpoints. The audio may be combined based on the audio level information. The conference audio may be provided to the plurality of conferencing endpoints.

Abstract: A telephone includes: a housing; a voice output device that is placed inside the housing and produces voice; a contact detection unit that detects a position of an object that contacts the housing; a position offset calculation unit that calculates a distance between the contact position of the object detected by the contact detection unit and the voice output device; and a voice adjustment unit that adjusts the voice produced from the voice output device, depending on the distance.

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: Embodiments of the present invention include methods and apparatuses for adjusting audio content when more multiple audio objects are directed toward a single audio output device. The amplitude, white noise content, and frequencies can be adjusted to enhance overall sound quality or make content of certain audio objects more intelligible. Audio objects are classified by a class category, by which they are can be assigned class specific processing. Audio objects classes can also have a rank. The rank of an audio objects class is used to give priority to or apply specific processing to audio objects sin the presence of other audio objects of different classes.

Abstract: The proposed echo canceller comprises:—an adaptive filter for receiving a signal affected by an echo, and for supplying a filtered signal that is an estimate of the echo;—a subtractor for subtracting this estimate from the received signal and supplying a residual signal;—means for detecting (7, 8) a ring back tone in said residual signal;—means for blocking (9) the received signal and replacing it by a locally generated ring back tone if a ring back tone is detected in the received signal;—a timer to determine a time period;—and means (11-17) for, during said time period, replacing the residual signal by some synthetic comfort noise when there is no ring back tone.

Abstract: An audio/video communication system displays the status of participants in a video chat session. The system includes multiple video chat capable (VCC) information handling systems (IHSs) that display video images of the participants. In this manner, each user may see the user's own video image as well as the video images of other users in the video chat session. When a user speaks, that user's VCC IHS detects audio, thus designating a speaker participant. This user's VCC IHS includes a gaze direction detector that determines at which particular user video image the user gazes, thus determining a target participant. The VCC IHS sends speaker participant ID information and target participant ID information to other VCC IHSs in the video chat session. In response, the other VCC IHSs display an indicator that designates one user video image as the speaker participant and another user video image as the target participant.