Abstract: A controller for an echo suppressor configured to suppress a residual echo of a far-end signal included in a primary error signal, the controller adapted for operation with a primary adaptive filter configured to form a primary echo estimate of the far-end signal included in a microphone signal and an echo canceller configured to cancel that primary echo estimate from the microphone signal so as to form the primary error signal, the controller comprising: a secondary adaptive filter configured to form a secondary echo estimate of the far-end signal comprised in the microphone signal; and control logic operable in at least two modes selected in dependence on a convergence state of the primary adaptive filter, the control logic being configured to control activation of the echo suppressor in dependence one or more transient or steady state decision parameters.

Abstract: A method implemented by a first network component (NC) in a digital subscriber line (DSL) system comprising retrieving an on-line reconfiguration (OLR) request from a control entity (CE), wherein the OLR request is generated by the CE based on a noise increase on a subscriber line between the NC and a second NC in the DSL system, and in response to the OLR request, transmitting a message to a receiver in the second NC to request for a reduced data rate on the subscriber line.

Abstract: A communication apparatus has an off-hook detection function, and has a first electric power supply for supplying electric power to a telephone control unit for controlling a telephone apparatus, and a second electric power supply for supplying electric power to an off-hook detection unit for detecting an off-hook of the telephone apparatus. The apparatus causes the off-hook detection unit to detect an off-hook of the telephone apparatus when, from a first power mode, a second power mode, in which electric power consumption is less than in the first power mode, is transitioned into.

Abstract: A method for processing an audio signal is described comprising receiving a first audio signal via a first receiving path comprising a first microphone; receiving a second audio signal via a second receiving path comprising a second microphone; and performing echo suppression of the first audio signal based on the first audio signal and the second audio signal.

Abstract: A Dual Tone Multi Frequency (DTMF) circuit for relaying DTMF digits includes a communication interface, a storage system configured to store a sampling window size, a DTMF test sequence, a returned DTMF sequence, and one or more DTMF digits, with the one or more DTMF digits to be transferred to a second DTMF circuit by the DTMF circuit, and a processing system. The processing system receives the one or more DTMF digits via the communication interface and an input sampling window, generates the DTMF test sequence and transmits the DTMF test sequence to the second DTMF circuit, compares the DTMF test sequence to a returned DTMF sequence received back from the second DTMF circuit, and if the returned DTMF sequence is not the same as the DTMF test sequence, adjusts the sampling window size of the input sampling window.

Abstract: In an example, time and frequency domain speech enhancement is implemented on a platform having a programmable device, such a PC or a smartphone running an OS. Echo cancellation is done first in time domain to cancel a dominant portion of the echo. Residual echo is cancelled jointly with noise reduction during a subsequent frequency domain stage. The time domain block uses a dual band, shorter length Adaptive Filter for faster convergence. Non-linear residual echo is cancelled based on an echo estimate and an error signal from the adaptive filters. A controller locates regions that had residual echo suppressed and which do not have speech and injects comfort noise. The controller can be full-duplex and operate non-linearly. An AGC selectively amplifies the frequency bins, based on the Gain function used by the residual echo and noise canceller.

Abstract: A method for acoustic echo cancellation in a communication device is provided that includes receiving a first near-end audio signal in the communication device, wherein the first near-end audio signal comprises acoustic echo of a far-end audio signal reproduced by the communication device, and performing echo cancellation on the first near-end audio signal to generate a second near-end audio signal with at least some of the acoustic echo removed, wherein the echo cancellation is performed responsive to presence or absence of double-talk (DT), and wherein a zero-crossing rate (ZCR) is used to detect the presence or absence of DT.

Abstract: While a dual tone multi-frequency (DTMF) signal can be easily processed if cleanly obtained, difficulties can arise if the DTMF signal is not cleanly obtained. In one example, part of the signal can be cleanly obtained, but part of the signal can be obstructed. In order to determine the content of the DTMF signal the parts that are cleanly obtained can be compared with known DTMF signals, such as DTMF signals that correspond with buttons on a telephone keypad. A classification of a closest matching known DTMF signal can be identified and that classification can be assigned to the partially obtained DTMF signal. Thus, without clearly obtaining the whole DTMF signal a content of the DTMF signal can be ascertained.

Abstract: A acoustic echo canceller (AEC) system may be configured to perform echo cancellation in the frequency domain. Features are disclosed for determining an estimated echo in the frequency domain using adaptive filters. An adaptive filter corresponding to a frequency bin can comprise a plurality of filter taps. Additional features are disclosed for updating the adaptive filter. In addition, a frequency-bin dependent step size controller may be used to control a step size used in updating the adaptive filters. Features are disclosed for determining the frequency-bin dependent step size.

Abstract: A digital subscriber line access multiplexer (DSLAM) configured to couple to a first subscriber line and a second subscriber line, wherein the DSLAM comprises at least one transmitter configured to transmit a first signal onto the first subscriber line in a first time slot, transmit a second signal onto the second subscriber line in the first time slot, wherein the first and second signals are linearly related and are intended for a first customer premise equipment (CPE), transmit a third signal onto the second subscriber line in a second time slot, and transmit a fourth signal onto the first subscriber line in the second time slot, wherein the third and fourth signals are linearly related and are intended for a second CPE.

Abstract: Methods and apparatus are provided for a managing at least one network address book of a user at a TS in a UPnP home network. A request to update a contact entry is received from a TCP that is coupled to the TS in the UPnP home network. The request includes contact information associated with at least one entity of the contact entry, and an identifier of a source of the contact entry. It is determined whether the identifier in the request is associated with any of one or more network address books in one or more WAN servers coupled to the TS via a WAN. When the identifier is associated with at least one of the one or more network address books, the contact entry in the at least one network address book is updated based on the contact information. When the identifier is not associated with any of the one or more network address books, the contact entry in a local address book in the TS is updated based on the contact information.

Abstract: The present invention provides a system for managing and disabling phone ring tone profiles, the system comprising: a. a storage system comprising (a) a plurality of ringtones adapted to be activated when an incoming call is made; and, (b) at least one list of identified callers; b. processor to execute at least one data processing system, said at least one data processing system comprising: i. an acquiring module operable to read an identifier of said incoming call or an incoming message; ii. a detecting module operable to detect if the identifier exists in said at least one list in the system, to determine the caller from said list, and to detect how many incoming calls have been made within a predetermined period of time; and iii. a disabling module adapted to either activate or disable said ringtones.

Abstract: A method and arrangement is proposed for adaptively generating a target noise margin for use on a DSL transmission line. The method includes determining a first quantity representing a current noise power on the line using line parameter data relating to the transmission performance of the line, determining a second quantity representing a worst-case noise power on said line using previously obtained values of the line parameter data, calculating the difference between the first and second quantities to generate a difference value, setting a target noise margin as at least equal to said difference value, and providing the target noise margin for use on the line upon initialisation of the line. In a preferred embodiment, the line parameter data is the bit loading on the line and the first and second quantities are values of the average signal-to-noise ratio on the line derived from the bit loading data.

Abstract: Systems and methods for removing noise from an audible signal are provided. More particularly, a vibration sensor is used to obtain a vibration signal from an environment including a communication device. The signal from the vibration sensor is combined with a signal from a microphone associated with the communication device, to create a modified audible signal. More particularly, a filtering or subtraction process can be performed with respect to the audible signal, at a time corresponding to an event detected as part of the vibration signal. The resulting modified audible signal can have reduced noise as compared to the original audible signal.

Abstract: Prediction of a channel capacity is accomplished based on a TDR echo without explicitly estimating the topology of the line. The prediction is based on obtaining a measured TDR echo, determining a theoretical TDR echo for a plurality of loop lengths, estimating the equivalent TDR length based on an optimization, updating the equivalent TDR length and utilizing the updated TDR length to predict one or more of the upstream and downstream data rates.

Abstract: A method and system for improving a perceived duplexity of handsfree telephone applications is disclosed. An echo suppression circuit for a device comprising loudspeaker and microphone is described. A circuit attenuates a subband of a transmit signal, wherein the transmit signal is captured by the microphone and wherein the transmit signal comprises an echo of a far-end signal rendered by the loudspeaker and a near-end signal. The attenuation circuit further determines a subband far-end indicator of a voice activity in the far-end signal; determines a subband near-end indicator of a voice activity of the near-end signal; determines a subband masking weight; determines a subband attenuation for the transmit signal in the subband; and attenuates the subband of the transmit signal using the determined subband attenuation.

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: An echo path change detector may be used to control the rate of adaptation in an acoustic echo canceller. When echo path change is declared, the rate of adaptation may be increased. However, echo path change should not be declared in the presence of double talk, because rapid adaptation during double talk is undesirable. Accordingly, various features are disclosed for detecting echo path changes while avoiding the declaration of such changes in the presence of double talk.

Abstract: Disclosed is a secure telephone call management system for authenticating users of a telephone system in an institutional facility. Authentication of the users is accomplished by using a personal identification number, preferably in conjunction with speaker independent voice recognition and speaker dependent voice identification. When a user first enters the system, the user speaks his or her name which is used as a sample voice print. During each subsequent use of the system, the user is required to speak his or her name. Voice identification software is used to verify that the provided speech matches the sample voice print. The secure system includes accounting software to limit access based on funds in a user's account or other related limitations. Management software implements widespread or local changes to the system and can modify or set any number of user account parameters.

Abstract: According to the invention, a user connection cable (TAL) that is to be associated with a user connection circuit (SLIC) and is used for connecting at least one user (TN) to at least one communication network is verified regarding switching on of a positive distribution voltage in the strands (a, b). The user connection circuit (SLIC) is initially switched to the silent interval mode and a test impedance (I) is switched between the strands (a, b) of the user connection cable (TAL), whereupon the current (i) flowing across the strands (a, b) and the voltage (u) to ground of a strand (b) are measured. Advantageously, the measurements are taken via measuring devices disposed on the user connection subassembly (SLMA).