Abstract: A system for providing augmented audio to users in a vehicle, comprising: a plurality of speakers disposed in a perimeter of a cabin of the vehicle; and a controller configured to receive a first audio signal and a second audio signal, to drive the plurality of speakers in accordance with a first array configuration such that a bass content of the first audio signal is produced in a first listening zone within the vehicle cabin, and to drive the plurality of speakers in accordance with a second array configuration such that a bass content of the second audio signal is produced in a first listening zone within the vehicle cabin, wherein in the first listening zone a magnitude of the first bass content is greater than a magnitude of the second bass content and in the second listening zone the magnitude of the second bass content is greater than the magnitude of the first bass content.

Abstract: A method and apparatus for classifying a spatial environment as open or enclosed are provided. In the method and apparatus, one or more microphones detect ambient sound in a spatial environment and output an audio signal representative of the ambient sound. A processor determines a spatial environment impulse response (SEIR) for the audio signal and extracts one or more features of the SEIR. The processor classifies the spatial environment as open or enclosed based on the one or more features of the SEIR.

Abstract: A system comprises automatic noise cancellation circuitry and interface circuitry operable to provide an interface via which a user can configure which sounds said automatic noise cancelling circuitry attempts to cancel and which sounds said automatic noise cancelling circuitry does not attempt to cancel. The interface circuitry may be operable to provide an interface via which a user can select a sound to whitelist or blacklist. The interface circuitry may be operable to provide an interface via which a user can increase or decrease an amount of noise cancellation that is desired. The interface circuitry may be operable to provide an interface via which a user can select from among three or more levels of noise cancellation.

Abstract: A portable communication device operating in a covert mode including an electronic processor that detects a presence of a wireless audio accessory. The portable communication device receives and transmits audio data to the wireless audio accessory. The electronic processor monitors a signal quality of the wireless link and a continued presence of the wireless audio accessory and determines whether the portable communication device is operating in a covert operation mode. Responsive to determining that the portable communication device is operating in the covert operation mode and determining, via the monitoring, that the signal quality of the wireless link has fallen below a threshold or that the wireless audio accessory is no longer present: receiving, at the portable communication device, subsequent audio data, recording the subsequent audio data, and refraining from outputting the subsequent audio data via a speaker of the portable communication device.

Abstract: Described is an echo cancellation method including buffering the at least one frame of a first voice signal, the at least one frame of the first voice signal including an echo signal, when the at least one frame of signal is preprocessed, separately obtaining, in a first timing period, the at least one frame of the first voice signal from a buffer and at least one frame of a reference signal matching the at least one frame of first voice signal from a reference signal queue, so that an estimated latency value, between each frame of first voice signal and a corresponding reference signal, remains within a preset range, performing echo cancellation processing on the echo signal in the at least one frame of first voice signal using the at least one frame of reference signal, to obtain a second voice signal, and outputting the second voice signal.

Abstract: The present invention relates to a voice signal processing method according to a state of an electronic device, and an electronic device therefor. An electronic device according to various embodiments comprises: a microphone; and a processor, wherein the processor can be configured to: obtain a voice signal using the microphone, check a state of the electronic device, and generate a first voice signal by filtering the voice signal using a first method, at least based on a determination of the electronic device being in a first state; generate a second voice signal by filtering the voice signal using a second method, at least based on a determination of the electronic device being in a second state; and transmit to an external electronic device a corresponding voice signal of the first voice signal or the second voice signal. In addition, other embodiments are possible.

Abstract: A system comprises automatic noise cancellation circuitry and interface circuitry operable to provide an interface via which a user can configure which sounds said automatic noise cancelling circuitry attempts to cancel and which sounds said automatic noise cancelling circuitry does not attempt to cancel. The interface circuitry may be operable to provide an interface via which a user can select a sound to whitelist or blacklist. The interface circuitry may be operable to provide an interface via which a user can increase or decrease an amount of noise cancellation that is desired. The interface circuitry may be operable to provide an interface via which a user can select from among three or more levels of noise cancellation.

Abstract: Disclosed are a method of recognizing an external device and an electronic device for processing the method. The electronic device may include a memory configured to store at least one earphone polarity information; and a processor electrically coupled to the memory, wherein the processor executing instructions stored in the memory is configured to detect an insertion of earphones into a connector of the electronic device, acquire a first polarity information stored in the memory, based on the first polarity information, set a sound path for inputting or outputting a sound to or from the electronic device, determine a second polarity information corresponding to the inserted earphones, determine whether the first polarity information is compatible with the second polarity information, and reset the sound path based on the second polarity information when the first polarity information is not compatible with the second polarity information.

Abstract: Audio processing method is described to continuously process an audio signal while changing between audio modes so that the listener does not experience a discontinuous sound during the transition. The first audio signal can be combined with the second audio signal to produce a smooth, continuous output signal that changes to only having the second audio signal over a time period. A global gain can be reduced during the transition from the first audio mode to a subsequent audio mode. Differences in the parameters from the first mode to the second mode can be determined and the parameters that differ are stepped from an initial value to a subsequent value over a transition time to produce a smooth audio signal.

Abstract: A mobile terminal including a wireless communication unit configured to provide wireless communication; a display on the terminal body, an acceleration sensor; a low-energy image sensor; and a controller configured to control the acceleration sensor to sense an acceleration of the mobile terminal, control the low-energy sensor to capture a surrounding image of the mobile terminal, in response to the sensed acceleration of the mobile terminal being related to a body movement of a user carrying the mobile terminal based on the surrounding image, output information related to the body movement of the user of the mobile terminal based on the sensed acceleration, and in response to the sensed movement of the mobile terminal being related to a movement different than the body movement of the user based on the surrounding image, disregard the sensed acceleration.

Abstract: There is provided an information processing apparatus including a matter extracting unit extracting a predetermined matter from text information, an action specifying unit specifying one or multiple action patterns associated with the predetermined matter, an action extracting unit extracting each of the action patterns associated with the predetermined matter, from sensor information, and a state analyzing unit generating state information indicating a state related to the matter, based on each of the action patterns extracted from the sensor information, using a contribution level indicating a degree of contribution of each of the action patterns to the predetermined matter, for a combination of the predetermined matter and each of the action patterns associated with the predetermined matter.

Abstract: [Object] To provide an information processing method, a control device, a recording device, an information processing system and a program which enable automatic control of a response sound of a recording device, with respect to an operation ordered by the control device. [Solution] Provided is an information processing method and an information processing system including establishing a connection, by a recording device, to a control device via communication, and determining a control mode of a response sound in the recording device with respect to an operation ordered by the control device depending on whether a plurality of recording devices are connected to the control device.

Abstract: The present disclosure relates to a microphone, and more particularly, to a microphone capable of improving a signal-to-noise ratio (SNR) by receiving a sound pressure through at least two acoustic sensing modules. According to an exemplary form of the present disclosure, a microphone includes first and second acoustic sensing modules including a diaphragm vibrated by a sound pressure introduced from the outside and a fixed membrane spaced apart from the diaphragm, and a signal processing module receiving first and second capacitance signals from each of the first and second acoustic sensing modules and removing and outputting noises included in the first and second capacitance signals based on the first and second capacitance signals.

Abstract: A method and system for a headset wireless charging dock, where the charging dock comprises a radio frequency (RF) radio, a charging induction coil, and a proximity sensor. The method may comprise sensing a presence of a headset using the proximity sensor, wirelessly charging a battery in the headset utilizing the charging induction coil, and wirelessly communicating commands, using the RF radio, to the headset to power down at least a portion of circuitry in the headset. The command may be communicated to the headset utilizing a protocol and a RF radio used by the headset to receive audio signals. The command communicated to the headset may power down audio processing circuitry in the headset. The charging induction coil may be inductively coupled to a coil in the headset to wirelessly charge the battery in the headset. The proximity sensor may comprise a Hall sensor.

Abstract: Methods and apparatuses are disclosed that allow an electronic device to autonomously adapt one or more user alerts of the electronic device. For example, some embodiments may include a method for operating a haptic device including driving a haptic device using a control signal, measuring a frequency related to the operation of the haptic device and comparing the measured frequency with a target frequency. A control signal is adjusted based on the comparison to drive the haptic device to the target frequency.

Abstract: Systems and methods for acoustic echo cancellation with wireless microphone and speaker systems are described herein. In one embodiment, an acoustic echo canceller is provided that can, among other things, cancel an acoustic echo component of an audio signal that is produced when an audio signal transmitted to a remote speaker on a lossy wireless link is picked up by a remote microphone and transmitted back to a base station on a lossy wireless link.

Abstract: A method and device are provided for the objective evaluation of voice quality of a speech signal. The device includes: a module for extracting a background noise signal, referred to as a noise signal, from the speech signal; a module for calculating the audio parameters of the noise signal; a module for classifying the background noise contained in the noise signal on the basis of the calculated audio parameters, according to a predefined set of background noise classes; and a module for evaluating the voice quality of the speech signal on the basis of at least the resulting classification relative to the background noise in the speech signal.

Abstract: A method for executing a menu in a mobile terminal and a mobile terminal thereof are disclosed, wherein the method includes displaying a first screen on a display unit of the mobile terminal, detecting whether an audio output device connection signal is generated by an external audio output device being connected to the mobile terminal while the first screen is displayed, and displaying a second screen including an audio output-related menu if the audio output device connection signal is generated.

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: 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: Embodiments of the present invention provide systems, methods, and computer-readable media for presenting a mute indicator to a caller when the caller is attempting to provide input to a call that has been muted. In particular, a determination is made that a mute function is engaged during a call. An area surrounding a calling system, such as a phone, is then monitored by a monitoring system to determine whether a caller is attempting to provide input to the call. A determination is then made that a caller is attempting to provide input to the call. The determination may be based on a caller providing a threshold volume of voice input within a threshold area of proximity near the calling system. A mute indicator is then presented to the caller to notify the caller that the call has a mute function engaged.

Abstract: The invention provides an apparatus capable of automatic volume adjustment. In one embodiment, the apparatus comprises a speaker, an array microphone located in the vicinity of the speaker, a beamforming module, a signal-to-noise ratio calculation module, and a volume adjustment module. The speaker first broadcasts a first audio signal. The array microphone converts a sound into a plurality of second audio signals. The beamforming module derives a speaker sound signal and an ambient noise signal from the second audio signals, wherein the speaker sound signal mainly comprises speaker sound components generated by the speaker and the ambient noise signal mainly comprises noises other than the speaker sound components. The signal-to-noise ratio calculation module calculates a ratio of a first power of the speaker sound signal to a second power of the ambient noise signal to obtain a signal-to-noise ratio.

Abstract: Audio processing systems, such as full duplex digital telephone systems, are provided in which analog automatic level control (ALC) circuitry is employed to automatically control and stabilize the microphone output audio signal levels to a constant level for optimal digital signal processing. For example, an audio communication device (10) includes a microphone (11) to generate an audio signal, an analog automatic level control (ALC) circuit (13), coupled to an output of the microphone (11), to control a level of the audio signal output from the microphone (11) and output a level-controlled audio signal, and a DSP (digital signal processing) circuit (14) to process the level-controlled audio signal output from the ALC circuit (13). The ALC (13) allows audio signals to be captured by the microphone (11) with wide dynamic range, while stabilizing the microphone output level within a reduced dynamic range compatible with the limited dynamic range of the DSP circuitry (14) for enhanced performance.

Abstract: In one embodiment, the present invention includes an apparatus having an automatic gain control (AGC) stage to receive an input signal from a communication channel physical medium, a first local gain stage coupled to an output of the AGC stage, an equalizer coupled to an output of the first local gain stage, an echo canceller to receive local data to be transmitted along the communication channel physical medium, and a second local gain stage coupled to an output of the echo canceller. Other embodiments are described and claimed.

Abstract: The invention relates to processing signal for improving a sound pick-up in telephone and/or videophone units. The inventive method consists in comparing (13) a signal transmitted by a unit microphone (Mic) with a threshold low value (THR_LOW) and in amplifying (14,15) said microphone-transmitted signal when it is less than the threshold low value. However, said steps depend on the test (10) carried out on a signal used for feeding the loud speaker (HP), in particular consisting in controlling said last signal (crete_HP) of a background noise order (BDFd) in such a way that the background noise is not amplified by an acoustical coupling between the microphone (Mic) and the loud speaker (HP).

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: Methods and arrangements for controlling modes of audio devices according to user selectable features are disclosed. The system can receive a user selection for an alternate mode for audio device. A monitor can monitor audio levels in a physical area and the system can automatically switch an audio device from a current mode to the alternate mode when the monitored audio level exceeds a predetermined threshold. The alternate mode can provide a higher audio level or volume than the current mode. The alternate mode can also be a closed caption mode, a record mode, and a transcript mode to name a few. Thus, when the ambient sound level reaches a predetermined level many audio devices can change modes based on user configured instructions.

Abstract: A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

Abstract: Even if an external output device like an earphone is disconnected during outputting of reproduced recorded contents, the recorded contents can be consecutively output at the time of subsequent reproduction. When an earphone connection detector detects that an earphone is disconnected during the reproduction operation of reproducing and outputting audio and image information in a recording section with the earphone being connected, a CPU performs control in such a manner as to stop the reproduction operation. When the earphone connection detector detects that the earphone is connected while the reproduction operation is stopped, the CPU resumes reproducing the audio and image information in the recording section at the stop position.

Abstract: Serializing circuitry is provided that can multiplex multiple device output signals and that can drive time-multiplexed data signals on the bus wires of a data path of an electronic system. Bus registers placed at the ends of the bus wires can register or buffer the data signals transmitted over the bus wires. The registered signals may be passed on to deserializing circuitry for demultiplexing the data signals to provide parallel device input signals. The bus registers and the serializing/deserializing circuitry can be provided along signal paths that require additional latency.

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 hybrid serial/parallel bus interface has a data block demultiplexing device. The data block demultiplexing device has an input configured to receive a data block and demultiplexes the data block into a plurality of nibbles. For each nibble, a parallel to serial converter converts the nibble into serial data. A line transfers each nibble's serial data. A serial to parallel converter converts each nibble's serial data to recover that nibble. A data block reconstruction device combines the recovered nibbles into the data block. The data block is employed by a gain controller. Each nibble has at least two start bits whose states collectively represent both a function and/or destination.

Abstract: Various embodiments of the present invention enable a system and method for reducing or entirely canceling background or environmental noise from a voice transmission from a communications device. A communications device, such as a mobile telephone, is configured with an environmental noise compensation, correction, or counterbalanced (correction) signal generator that is connected between at least one microphone and a continuous time quadrant multiplication. Optional discrete time or digital signal processing may be applied. The original output of the at least one microphone and a compensation, correction, or counterbalanced or counter-balancing signal generated by the environmental noise compensation, correction, or counterbalanced signal generator are mixed or otherwise combined together after being received by the antenna to the receiver.

Abstract: A high-speed bit stream interface module interfaces a high-speed communication media to a communication Application Specific Integrated Circuit (ASIC) via a Printed Circuit Board (PCB). The high-speed bit stream interface includes a line side interface, a board side interface, and a signal conditioning circuit. The signal conditioning circuit services each of an RX path and a TX path and includes a limiting amplifier and a clock and data recovery circuit. The signal conditioning circuit includes an AGC loop, an equalizer, and an equalizer feedback loop. The AGC loop includes a gain path and a feedback path that couples to the output of the equalizer. The equalizer feedback loop couples to the output of the equalizer and produces spectral shaping control settings that the equalizer uses to produce an equalized high-speed serial bit stream at an equalizer output.

Abstract: A voice recognition system includes an adaptive filter and a subtractor. The adaptive filter generates a simulated talk-back voice y(n) by setting a filter coefficient simulating a transfer system in which an input voice corresponding to a voice command and a talk-back voice output from a speaker are input into a microphone and by filtering a talk-back voice x(n). The subtractor extracts the input voice by subtracting the simulated talk-back voice y(n) from mixed sound input into the microphone. With this configuration, the talk-back voice is attenuated from the mixed sound including the input voice and the talk-back voice input tedinto the microphone, and then, the mixed sound is supplied to a voice recognition engine. Accordingly, the user can input his/her voice during a talk-back operation without the need to interrupt it by pressing a speech button every time the user wishes to input the voice. The voice recognition operation time can be thus reduced.

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 system comprising processing logic and sound-capturing logic coupled to the processing logic. The sound-capturing logic provides a captured signal to the processing logic. The captured signal is associated with a property. Transceiver logic is coupled to the processing logic. The transceiver logic provides a received signal to the processing logic. The received signal is associated with a volume. Using a compression technique, the processing logic adjusts the volume in accordance with the property.

Abstract: A phone call mute notification applies an aural effect to the user's speech, other parties' speech, non-speech, or combinations thereof. The phone call mute notification does not render speech unintelligible and may provide a constant indication that the phone call mute function is active. Furthermore, the phone call mute notification does not serve as a distraction to the user, because the aural effect may be applied to speech itself rather than drawing the attention of the user away from the speech.

Abstract: A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

Abstract: A circuit where the same amplifiers and the same volume adjustment circuitry are used for the ringer mode as well as for the audio mode of a mobile phone is achieved. The volume adjustment in the audio and ringer mode is provided by a precise amplitude setting via the gain control stages of an inverting voltage amplifier used in a bridge circuit through a current-voltage conversion. This volume adjustment circuit avoids the high power dissipation of a volume control through pulse width modulation (PWM) and avoids the risk of over-and undershooting of the amplifier's output signal due to a high slew rate in combination with the inductance of the loudspeaker. The current-voltage conversion is performed by a series of resistors activated by a series of correspondent switches. High impedance current less sense paths are eliminating the parasitic effect of the resistance of low cost standard switches to adjust the volume of the loudspeaker in the audio and the ringer mode.

Abstract: An electronic device comprises an earpiece speaker, a loudspeaker, and a processing circuit. The processing circuit is configured to receive a wireless telephony audio signal, to separate the audio signal into a first portion and a second portion, the first portion having higher frequency components than the second portion, wherein the processing circuit is configured to provide the first portion of the audio signal to the earpiece speaker and the second portion of the audio signal to the loudspeaker.

Abstract: Embodiments of the method and system provide for automatic balancing of audio signal strength levels on voice paths between first and second parties in a telecommunication system. The method in an embodiment may have the steps of: checking for a subscription to audio signal strength level balancing by at least one of the first and second parties; and balancing, if at least one of the first and second parties has the subscription, audio signal strength levels on voice paths between the first and second parties. The system implements the method.

Abstract: A device and method for handling announcement of an incoming call in a telephone handset at least in part in accordance with a measured ambient noise level is provided. In some embodiments, the disclosed device includes an audio signaling mechanism such as a ringer whose loudness is automatically adjusted in accordance with a measured ambient noise levels. Alternatively or additionally, the present invention provides an audio speaker for outputting received voice communications, wherein the loudness of the speaker is determined at least in part in accordance with a measured ambient noise level. In some embodiments, the present invention provides methods and devices for handling announcement of an incoming call at least in part in accordance with electrical output from other sensing circuits such as location sensing circuits and environmental circuits.

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: Duplex communication methods and systems with maximum gain limiting such as in telephone handsets and speaker phones are disclosed. The communications system such as a telephone set may generally include a digital signal processor (DSP) having a receive and transmit path in communication with a receiver and a transmitter, respectively, a volume amplifier for amplifying signals on the receive path, and a maximum gain limiter for determining a total return loss between the receive and transmit paths and for limiting a volume amplification level of the volume amplifier to a maximum depending on the total return loss. The maximum gain limiter determines the total return loss by maintaining a total loop gain for the telephone set at less than 0 dB to prevent unstable operation and howling.

Abstract: A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with umodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

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.