Abstract: A data slicer for converting an envelope signal of an amplitude-modulated wave into a binary signal, comprises: an average level generation circuit configured to generate an average level of the envelope signal by averaging the envelope signal per time; a fixed voltage value generation circuit configured to generate a fixed voltage value; a reference level generation circuit configured to generate a reference level in accordance with the fixed voltage value and the average level of the envelope signal; and a comparison circuit configured to compare a signal level of the envelope signal with the reference level to output a result of the comparison as the binary signal.

Abstract: Examples provide an adaptation circuit and apparatus, method and computer programs for adapting, fabricating and operating, a radio transceiver, a mobile transceiver, a base station transceiver and storage for computer programs or instructions. The adaptation circuit (10) is configured to adapt a local oscillator signal in a radio transceiver (30). The radio transceiver (30) comprises a transmission branch (14) and a reception branch (16), which are subject to cross-talk. The reception branch (16) comprises a local oscillator (18) configured to generate the local oscillator signal. The adaptation circuit (10) comprises a control module (12) configured to determine crosstalk level information between the transmission branch (14) and the reception branch (16), and to adapt the local oscillator signal based on the crosstalk level information.

Abstract: This applications relates to methods and apparatus for amplifying signals from capacitive transducers, in particular MEMS transducers such as MEMS capacitive microphones. An amplifier circuit has a signal node for receiving the input signal, a transducer biasing node for outputting a transducer bias voltage for biasing the capacitive transducer, and a voltage buffer configured to generate a buffered bias voltage at a buffer node. An amplifier arrangement is configured to receive the input signal from the signal node and the buffered bias voltage.

Abstract: An amplifier circuit has a transducer biasing node for outputting a transducer bias voltage for biasing the capacitive transducer and a signal node for receiving the input signal. An amplifier arrangement comprising a feedback resistor network provides an amplified output signal. A voltage buffer provides a buffered bias voltage at a buffer node which is connected to a terminal of the feedback resistor network, to at least partly define the quiescent level of the output signal. The buffer node is electrically coupled to the transducer biasing node via a capacitance which may form part of a bias filter.

Abstract: An electronic device may include a housing and a directional microphone. The housing may include a front surface, a rear surface positioned opposite the front surface, and a hollow cavity positioned between the front surface and the rear surface. The hollow cavity may include a front opening defined in the front surface and a rear opening defined in the rear surface. The directional microphone may include a front port and a rear port. The directional microphone may be mounted in the hollow cavity of the housing with the front port oriented toward the front opening and with the rear port oriented toward the rear opening.

Abstract: This document discusses among other things apparatus and methods for protecting circuit elements from harmful voltages. In an example, an apparatus can include an amplifier configured to receive an input signal and to provide an estimate of a first output signal, a peak detector to receive the estimate and to generate a comparison signal that is active when the amplified input signal exceeds a threshold value, and a timer configured to activate a second output signal if the comparison signal is active for at least a selected time period. The timer can include a first digital input and the selected time period can be set using a state of the first digital input.

Abstract: Disclosed are various embodiments for providing a power-efficient driver architecture supporting rail-to-rail operation in full duplex mode. A driver is configured to drive a duplex signal over a transmission medium. A hybrid is configured to recover a received signal from the duplex signal. The received signal is generated by a remote transceiver. The driver is configured to drive the duplex signal based at least in part on the received signal recovered by the hybrid.

Abstract: Method and apparatus for microphone matching for wearable directional hearing assistance devices are provided. An embodiment includes a method for matching at least a first microphone to a second microphone, using a user's voice from the user's mouth. The user's voice is processed as received by at least one microphone to determine a frequency profile associated with voice of the user. Intervals are detected where the user is speaking using the frequency profile. Variations in microphone reception between the first microphone and the second microphone are adaptively canceled during the intervals and when the first microphone and second microphone are in relatively constant spatial position with respect to the user's mouth.

Abstract: A system for estimating the occlusion effect comprises a hearing aid adapted for being set up for operation in an occlusion measurement mode. The hearing aid comprises a first transducer (9) for transforming an acoustic sound level external to a hearing aid user's ear canal into a first electrical signal. The hearing aid comprises a second transducer (10) for transforming the acoustic sound level in the occluded ear canal into a second electrical signal. The system comprises a filter bank (21, 22) for splitting the first and the second digitized electrical signals into a first and a second band split digitized electrical signal, respectively and the estimated sound pressure must be a factor ?{square root over (r)} larger than the estimated leakage when deciding if the estimated sound pressure in a given band can be applied in the calculation of the occlusion effect. The invention further provides a method for estimating the occlusion effect.

Abstract: A selector selects an analog audio signal input to one input port from among analog audio signals input to multiple input ports according to an instruction from the user. An analog gain control circuit amplifies the analog audio signal received from the selector, with a corresponding one of the gains set for the respective input ports. An analog gain control circuit is configured to gradually change its gain when the gain is switched. An A/D converter converts an output signal of the analog gain control circuit into a digital audio signal. A first audio signal processing circuit is monolithically integrated on a signal semiconductor substrate.

Abstract: A method of reducing noise in an acoustic system, the method comprising at a first user terminal: receiving an audio signal from at least one further user terminal over a network; executing a communication client on a processing unit, the client configured so as when executed to: supply the audio signal to an audio signal processing module of the first user terminal, wherein the processing module processes the audio signal, whereby a level of gain is applied to the audio signal, and outputs a processed audio signal to a speaker; estimate a noise level of the audio signal and the processed audio signal and estimate the gain applied by the processing module taking into account both the noise level estimates; selectively apply a system gain reduction step to at least one of the audio signal and an audio signal received via a microphone, based on at least the estimated gain.

Abstract: A self configuring headset system comprising an interface unit, a method of setting up a headset system to a telephone device using and interface unit and a telephone interface unit is provided.

Abstract: There is provided a signal processing apparatus including a recognition section configured to recognize at least one of a state of a first user, a state of a second user, and a relationship between the first user and the second user, a signal processing unit configured to perform signal processing on voice information corresponding to voice articulated by the first user, or voice information corresponding to voice that is transmitted from a remote location and articulated by the second user, and a control unit configured to control the signal processing unit to perform signal processing on at least one of the voice information corresponding to the voice articulated by the first user, and the voice information corresponding to the voice that is transmitted from the remote location and articulated by the second user, in accordance with a recognition result of the recognition section.

Abstract: An adaptive hybrid system is coupled to a loop for adjusting trans-hybrid loss. The system comprises a fixed portion comprising a first receiver transfer function block and a first hybrid transfer function block. The fixed portion is configured to receive a far-end signal and mitigate frequency dependent attenuation experienced by the far-end signal. The system also comprises a variable portion comprising a second receiver transfer function block and a second hybrid transfer function block configured to subtract a transmit echo from the received far-end signal.

Abstract: A multi-handset telephone system has a first telephone handset having an antimicrobial agent situated thereon, a first amplifier that provides built in hearing impaired support, and a first integrated volume control actuator. Further, the multi-handset telephone system has a second telephone handset having the antimicrobial agent situated thereon, a second amplifier that provides built in hearing impaired support, and a second integrated control actuator. The second telephone handset is operably connected to the first handset through a handset cord such that a first user speaks on the first handset and the second user speaks on the second handset to have a face-to face conversation with each other at the same time as having a conversation with a third party remotely.

Abstract: Systems and methods for managing the volume of multiple VoIP streams are disclosed. The system includes a VoIP server configured to receive an input audio stream from a first VoIP handset, create separate output audio streams from the input audio stream for transmission to second and third VoIP handsets, and to connect to a communications network. The system also includes a volume control table coupled to the VoIP server, the volume control table including records of volume adjustments made during prior conversations between the two or more VoIP handsets. The VoIP server is further configured create the separate output audio streams such that one or more of the output streams has a volume that is different than input audio stream based on the records.

Abstract: The present invention discloses a circuit with three-stage of power-on sequence used for suppressing the pop noise in audio system. It mainly comprises a first resistor (R1); a capacitor (Cout); a first switch (SW1); a second switch (SW2); a soft start device; a first feedback amplifier; and a second feedback amplifier. By using the three-stage of power-on sequence, the present invention can effectively suppress the pop noise when the audio driver is power on.

Abstract: A method for through active-silicon via integration is provided. The method comprises forming an electrical device in a handle wafer. The method also comprises forming an isolation layer over the handle wafer and the electrical device and joining an active layer to the isolation layer. Further, the method comprises forming at least one trench through the active layer and the isolation layer to expose a portion of the handle wafer and depositing an electrically conductive material in the at least one trench, the electrically conductive material providing an electrical connection to the electrical device through the active layer.

Abstract: Systems and methods for a line amplifier circuit are disclosed. The line amplifier circuit generally includes a first microphone transmit line, a second microphone transmit line, and a light emitting diode with a controllable variable light output. A photo-resistor cell is coupled across the first microphone transmit line and the second microphone transmit line. The photo-resistor cell is operable to attenuate gain on the first microphone transmit line and the second transmit line responsive to the controllable variable light output from the light-emitting diode.

Abstract: A method is disclosed that enables the managing of the overall sound level in an enterprise environment where telephones are used. A data-processing system such as a private branch exchange monitors whether one or more telephones are in use. Based on detecting when a first endpoint is in use and, therefore, when the endpoint's user is present, the private branch exchange controls one or more characteristics of the loudspeaker volume at a second endpoint. By accounting for other considerations such as the spatial closeness between the endpoints, which can be determined from office dimensions stored in a database, the private branch exchange of the illustrative embodiment is able to determine the degree of sound that is coupling over from one endpoint location to another. On a larger scale, the exchange is able to control the loudspeaker volumes of all of the endpoints in the workplace area. In doing so, the exchange manages the overall acoustic noise present.

Abstract: An echo canceller apparatus comprises a receive side attenuator coupled in a receive side signal path that is configured to couple from a conference call bridge to a caller; a convolution processor coupled to the receive side signal path at a convolution processor pick-off point; a double-talk detector coupled to the receive side signal path and to a sending side signal path that is configured to couple from the caller to the conference call bridge; and logic coupled to the receive side attenuator which when executed is responsive to a double-talk condition detected by the double-talk detector and operable to determine a level of echo canceled by the convolution processor, to determine an additional amount of attenuation to introduce, and to activate the receive side attenuator to introduce the additional attenuation.

Abstract: A method for improving the intelligibility of an incoming telephone signal, including boosting loudness of at least one band of poorly heard frequencies of the signal within at least one band of intensities of the signal, the band lying below a predetermined intensity level at which telephone standard conformance testing is performed, thereby to generate a differentially boosted telephone signal.

Abstract: Driver circuit for driving a transmit signal for a line comprising a signal amplifier which delivers a transmit signal, present at a signal input of the signal amplifier, amplified with a gain to a signal output, a line interface circuit which is connected between the signal output of the signal amplifier and the line, and comprising a hybrid circuit which is connected to the line interface circuit, wherein the hybrid circuit suppresses the transmit signal and couples a received signal received via the line to a signal input of the signal amplifier for the purpose of impedance synthesis.

Abstract: A system includes a plurality of modules which are configured for determining a suitable impedance for a modem. The plurality of modules send signals with a plurality of signal frequencies to a telephone and receive feedback signals from the telephone under each of a plurality of impedances of the modem, compute a total decibel value of the feedbacks under each impedance, and determine a suitable impedance according to a minimum total decibel value from the plurality of total decibel values.

Abstract: Methods, systems, and devices are provided for controlling the power level of voice signal streams. A gain controller embodiment includes a measurement module, a gain factor setting module, and a gain adjustment module. The measurement module includes program instructions to measure a power level of a voice signal stream. The gain factor setting module includes program instructions to set a gain value by comparing the measured power level to a threshold. The gain adjustment module includes program instructions to adjust a power level of a voice signal stream by applying the gain value to the voice signal stream.

Abstract: A communication device may determine total gain for a downlink processing path via the communication device. The total gain may be determined based on determination of audio requirements for each of a plurality of audio processing devices supported via the communication device, during the downlink processing path. The communication device is operable to determine a default mix signal gain, a downlink gain adjustment and a calibration gain based on the determined audio requirements, and to calculate the total gain for the downlink processing path based on the determined default mix signal gain, the downlink gain adjustment and the calibration gain. The determination of the total gain may be based on determination of a telephony state in the communication device. The calibration gain may be determined during calibration of the communication device, whereas the downlink gain adjustment may be calculated based on an adjustment step-size and/or a mode based gain index.

Abstract: The present invention overcomes interface problems between proprietary handset ports on telephone base units and voice/data accessory products by allowing a user to automatically calibrate the telephone accessory product for an optimal interface match with the intended telephone base unit. This is accomplished through the use of a “Smart Interface Technology” (SIT) integrated chip set consisting of a full custom analog and semi-custom digital integrated circuit. The SIT incorporates three different methods for “learning” the characteristics of 4-wire port modular interfaces found in all telephone station sets. Basically, these methods determine the appropriate 4-wire terminal configurations, the transmit and receive channels of the intended telephone base unit, and adjust the channel sensitivities until an optimal and clear signal is provided for the user.

Abstract: at least one self oscillating loop circuit which oscillates with a switching frequency (fs), wherein the self oscillating loop circuit (18, 19, 20) comprises an output driver (19) which is switched with said switching frequency and a loop filter (20) which feeds back an amplified loop signal output by said driver (19) to an input of said self oscillating loop and a trimming circuit (21) for trimming the switching frequency (fs) of said self oscillating loop circuit.

Abstract: A system, method, and apparatus are directed towards managing an audio message, such as a Voice over Internet Protocol (VOIP) message over a network. The invention employs a statistical mechanism to automatically optimize a gain control for setting a volume of an audio message being sent by a client device. An initial gain value is automatically adjusted based, in part, on a statistical sampling of energy levels in the audio message. Environmental factors, such as a sound card within the client device, background noise, and the like, may also be considered through a setting of a servo coefficient that may be used to map between volume levels and decibel levels. The servo coefficient may also be adjusted based, at least in part, on decibel (dB) feedback information from a destination device for which the audio message is intended.

Abstract: A method and apparatus for controlling the transmit volume level of a speakerphone includes receiving a transmit audio signal. If the speakerphone is in a speakerphone mode, the method further includes measuring a send output level of the transmit audio signal, comparing the send output level of the transmit audio signal with a send output level threshold value, and controlling the send output level of the transmit audio signal based upon the comparison of the send output level of the transmit audio signal and the send output level threshold value using automatic gain control. Some embodiments further include determining the send output level threshold value based upon a send loudness rating (SLR) target difference value between a speakerphone SLR target value and a handset SLR target value. In some embodiments, if the speakerphone is in a handset mode, the send output level of the transmit audio signal remains substantially the same.

Abstract: A method and a device for amplifying a reception signal are provided, involving the adjustment of a gain by which the reception signal is amplified, in particular as a function of a signal level of the reception signal. For example, lowpass filtering with a variable cutoff frequency is used to filter out a gain-dependent frequency range of the reception signal, which may especially contain an undesirable echo signal. Such a method and such a device are suitable in particular for the reception of ADSL signals in a central office.

Abstract: An electric circuit for amplifying an xDSL signal comprises an operational amplifier and a signal monitoring circuit. The operational amplifier is configured to amplify the xDSL signal, is powered by a variable voltage supply and has a gain which is adjustable by an operating mode control signal. The signal monitoring circuit is activated by the operating mode control signal and is configured, when activated by the operating mode control signal, to generate a control signal to adjust the voltage of the variable voltage supply in order to adjust the maximal signal swing of the output signal of said operational amplifier. The control signal is generated by comparing the amplitude of the xDSL signal applied to the electric circuit with an amplitude threshold value.

Abstract: This device for processing audio signals for telecommunication equipment comprises automatic gain control means (18) acting on the level of the received audio signal (Sr) by amplifying and attenuating the signal as a function of the level of the received signal and echo cancelling means (22). The automatic gain control means (18) comprise amplification means (19-b) and attenuation means (19-a) for separately amplifying and attenuating the received signal downstream and upstream of the echo cancelling means (22), when considering the propagation direction of the received signal.

Abstract: A multi-handset telephone system has a first telephone handset having an antimicrobial agent situated thereon, a first amplifier that provides built in hearing impaired support, and a first integrated volume control actuator. Further, the multi-handset telephone system has a second telephone handset having the antimicrobial agent situated thereon, a second amplifier that provides built in hearing impaired support, and a second integrated control actuator. The second telephone handset is operably connected to the first handset through a handset cord such that a first user speaks on the first handset and the second user speaks on the second handset to have a face-to face conversation with each other at the same time as having a conversation with a third party remotely.

Abstract: A multi-handset telephone system includes is a first-base that includes circuitry for communicating with an external telephone line, a ringer for signaling when an incoming call is received, and a cradle for holding a single handset, a first handset, and a handset connection port. A second base has a cradle and a second handset. The interconnection is a flexible hardwired connection, and there is a switch associated with the second base. A port communicates the signal from the incoming call, and an outlet splits the telephone signal from the line into a plurality of handset cords. A first handset is connected with a first handset cord and a second handset is connected with a second handset cord of the plurality of handset cords. The corresponding plurality of handset cords permits a plurality of users in close proximity to speak on the plurality of handsets in a face-to-face situation with a third party. The telephone system is a digital and the second base does not have a ringer.

Abstract: A method (500) and an apparatus (300) are presented for processing a signal in a receiver in accordance with Ultra Wideband (UWB) protocol. The received signal is acquired during a first preamble portion (411) of a data frame associated with the UWB protocol. During acquisition a first Automatic Gain Control (AGC) operation (606) is performed associated with a noise component of the signal. A second AGC operation (608) associated with a pulse component of the signal is performed after the acquiring the signal during a second preamble portion (413).

Abstract: A modem includes a base unit for transmitting a data signal, and a communication card which receives the data signal from the base unit over a wireless medium and which performs echo canceling on the data signal. The base unit is in communication with a telephone line and receives an original signal from the telephone line. The base unit generates an RF modulated signal based on the original signal. The base unit includes a transmitter for transmitting the data signal. Circuitry in the base unit receives the original signal from the telephone line and generates a combined data signal from the original signal and echo signals and maintains a peak voltage of the combined signal which is within the linear amplification region of the transmitter.

Abstract: A method and system for managing a communication session is provided. The communication session is associated with multiple communication devices. The method includes learning (304) a set of derived acoustic features of an audio communication signal that is associated substantially only with one user of a communication device. The method also includes receiving (306) a communication session signal. The communication session signal is an audio signal that includes a combination of audio communication signals. Each audio communication signal of the audio communication signals is associated with a user of a communication device of the multiple communication devices. The method includes modifying (308) the communication session signal based on the set of derived acoustic features.

Abstract: A digital communication system includes a transmitter and a receiver and communicates on telephone line. Receipt gain of the digital communication system is controlled in accordance with the following. Initialization signals generated at the transmitter and transferred through the hybrid circuit on the telephone line are received for a predetermined time period. An average magnitude value of the initialization signal received during the time period is compared with a reference magnitude value. Receipt gain is controlled according to the difference between the average magnitude value of the receipt signal and a reference value. In this manner, when the system performance is changed according to permissible tolerance of the analog discrete components used in the ADSL system, gain of the gain amplifier is automatically controlled. Thus, the ADSL system can operate with optimal performance under given conditions.

Abstract: A method, apparatus, system, and signal-bearing medium that in an embodiment determine a degree of correlation between a speaker output signal and a microphone input signal and modulate an adaptive gain of an acoustic echo canceller based on the degree of correlation.

Abstract: A voice signal clipping circuit comprising, an amplifier which amplifies an input voice signal to a permissible level and a clipper which clips an amplitude component in excess of the permissible level from an output signal from the amplifier.

Abstract: A communications device having a user configurable speakerphone feature is provided. A user may be provided with a plurality of selectable settings for configuring the operation of a microphone or the processing of speech. For example, a user may select settings to filter ambient noise for a cleaner sound or increase a microphone's sensitivity in a conference call environment. In one embodiment, the user may select a setting that changes a microphone level or microphone operating range. When the microphone detects speech, the speech may be processed in accordance with user selected processing settings. For example, a user may select settings to specify a gain control, a frequency response, or an activity threshold. Thus, a user may configure a speakerphone to improve sound quality on the receiving end.

Abstract: A digital telephone answering device having speakerphone capability allows a recorded message to be played back and heard by the far-end party as well as over the local speakerphone by the near-end party during speakerphone operation as if it were a normal receive signal. Moreover, normal speakerphone conversation is possible during this conversational playback mode allowing the far-end party and/or near-end party to break-in over the played back pre-recorded message as desired and be heard by the other party. Both message and conversation signals are preferably (but not necessarily) at similar levels. Also, the message playback at the near end is preferably (but not necessarily) subject to the same speakerphone digital volume control as that received at the far end.

Abstract: The present invention is directed to a telecommunication device, usable with standard telecommunication equipment, that automatically resets one or more acoustic parameters to predetermined levels prior to the initiation of the next communication session.

Abstract: An apparatus for enhancing audibility of a far-end speech signal from a far-end user to a near-end user in a telephony system includes a near-end background noise signal level estimator (26) at the near-end user. A near-end speech signal level estimator (24) is also provided at the near-end user. A gain control logic (22) determines a gain (G) for amplification of the far-end speech signal based on both estimated speech and background noise signal levels.

Abstract: A push-button signal receiving device and a gain control method whereby error in validity determination of a received signal that accompanies the insertion of an AGC circuit can be prevented. An input bush-button signal is amplified by amplifying circuit and separated into signals of low- and high-frequency bands by first and second band-pass filters, respectively. Dialed number identifying circuit detects the frequencies of the separated signals to identify the dialed number, determines validity of the received push-button signal, and outputs the identified dialed number in accordance with the result of validity determination. Level detecting circuit detects the level of the output signal from the amplifying circuit. Gain control circuit controls the amplification gain of the amplifying circuit in accordance with the detected signal level, and holds the value of the amplification gain of the amplifying circuit while the received push-button signal is judged valid by the dialed number identifying circuit.

Abstract: A system and method for adjusting the volume level of a communications device in response to ambient noise. In one embodiment, ambient noise is sampled once and the volume of a speaker associated with the communications device is increased to, and maintained at, a level sufficient to overcome the ambient noise such that a user can easily carry on a conversation or hear what is being transmitted. In another embodiment, ambient noise is periodically sampled and the volume of the speaker is adjusted in response to the sampled ambient noise. The system and method provides enhanced user convenience and power saving advantages.

Abstract: The present invention relates to a loudspeaker volume control arrangement for a telephone having a loudspeaker and a microphone which controls the loudspeaker volume of the telephone based on the estimated distance between the microphone and the loudspeaker.

Abstract: A system and method for reducing or entirely canceling background or ambient noise from a voice transmission from a communications device. A communications device, such as a mobile telephone, is configured with an ambient noise compensation signal generator that is connected between a microphone and a mixer. The original output of the microphone and a compensation signal generated by the ambient noise compensation signal generator are mixed together prior to being passed to a transmitter. In one embodiment a buffer is provided between the microphone and the mixer to help synchronize the timing of the signals to be mixed. In another embodiment a second microphone is employed to detect ambient noise.

Abstract: An apparatus for transmitting charging signals on a data transmission path having a line impedance includes a driver device for setting a line voltage level that corresponds to the line impedance. The driver device is configured to provide, in response to a driver input voltage level, a driver output current and a driver output voltage level. A current detection unit provides a current signal indicative of the driver output current and a current signal matching unit receives the current signal from the current detection unit and adjusts a level of the current signal for further processing by a filter device and by a regulation device. A matching filter unit then matches the driver device to a data transmission path unit.