Abstract: The present disclosure relates to acoustic technology, in particular to an earphone including a sound generation portion. The sound generation portion includes a transducer and a housing for accommodating the transducer. The earphone further includes an earhook. The earhook includes a first portion and a second portion. The first portion may be hung between an auricle and the head of a user, and the second portion may be connected to the first portion, extends toward an anterolateral side of the auricle, and may be connected to the sound generation portion. The sound generation portion may be fixed near an ear canal without blocking an opening of the ear canal. In at least one frequency range, when an input current of the transducer does not exceed 35.3 mA, a maximum sound pressure that the sound generation portion is able to provide to the ear canal may not be smaller than 75 dB.

Abstract: Embodiments of the present disclosure provide apparatus, methods and computer programs for ultrasonic proximity-detection.

Abstract: A terminal device includes: a first communicator that carries out short-distance radio communications with a terminal device; a second communicator that carries out long-distance radio communications with the terminal device; a detector that detects an identifier, for identifying the terminal device, from a reception signal received by the first communicator; a determiner that determines signal strength of the reception signal detected by the detector; a storage that stores information relating to the terminal device and the signal strength; a storage controller that exercises such control that the identifier and the signal strength are correlated with each other and stored in the storage as a proximity station list; a setter; and an output controller.

Abstract: Embodiments discussed herein relate to automatic communication management, such as automated closure of a communication in circumstances where establishment of the communication is accidental, particularly from mobile terminals. A method of use of automatic communication management is disclosed. The method can include a triggering, following establishment of a communication between at least a communication terminal of the user and at least one other communication terminal, an activated automatic communication management which enables triggering of a closure of the established communication in progress as a function of the detected vocal sound level on at least one audio stream of the established communication in progress. In this way, if establishment of the communication is detected as being unintentional because of the vocal sound level of the communication, the communication will be closed.

Abstract: The disclosure provides an audio signal processing circuit and an audio signal processing method, adapted for processing the input signal of a speaker configured with a rated power. The audio signal processing circuit includes, but not limited to, an audio signal generator and a power adjuster. The audio signal generator provides an audio signal including a high-frequency signal and a middle and low-frequency signal. The power adjuster is electronically coupled to the audio signal generator, and adjusts the power of the high-frequency signal according to an enhanced power larger than the rated power without adjusting the power of the middle and low-frequency signal. The output signal of the power adjuster can be inputted into the speaker. Accordingly, the transmission range for the high-frequency signal can be improved.

Abstract: A method and system of conferencing can include the steps of initiating a conference call at a communication device with two or more communication devices and selecting to suppress a voice communication of at least one communication device on the conference call where a modified electronic signal is generated with the selected at least one communication device so that the voice communication from the selected at least one communication device is inaudible. The method or system further includes sending the modified electronic signal to at least one other communication device on the conference call. Other embodiments are disclosed.

Abstract: A computing system, computer-implemented method, and computer program product are provided for facilitating behavior based notifications concerning multiparty communication threads. According to an implementation, the computer-implemented method may include defining a rules set associated with a user for participation in a multiparty communication thread. The method may also include evaluating a plurality of communications associated with the multiparty communication thread. The method may also include determining when a given communication associated with the multiparty communication thread satisfies one or more of the rules included within the rules set. The method may further include delivering the given communication based upon, at least in part, a determining that the given communication satisfies the one or more rules.

Abstract: Devices and techniques are generally described for control of a voice-controlled device using acoustic echo cancellation statistics. A reference signal representing the audio stream may be sent to an acoustic echo cancellation (AEC) unit. A microphone may receive an input audio signal and send the input audio signal to the AEC unit. The AEC unit may attenuate at least a part of the input audio signal. AEC statistics related to the attenuation of at least the part of the input audio signal may be determined over a first period of time. A wake-word in the input audio signal may be detected during the first period of time. A determination may be made that the wake-word is part of the playback of the audio stream based at least in part on the AEC statistics.

Abstract: A computer-implemented method to select a user profile is disclosed. The method may include obtaining, from a first device, a communication attempt at a user device. The method may further include obtaining first device data associated with the communication attempt and with the first device. The method may also include comparing the first device data with multiple data elements of each of multiple user profiles associated with the user device. The method may further include selecting a user profile of the multiple user profiles that includes the data element that matches the first device data in response to the first device data matching a data element of the multiple data elements. The method may also include loading an audio profile for the selected user profile on the user device.

Abstract: A TDD radio unit and a real-time RX supervision method used therein are provided. The method comprises steps of: setting the radio unit into an RX supervision status in a TX timeslot, wherein the setting comprises: connecting a TX observation module in the radio unit to a termination port of a Transmit/Receive (TR) switch in the radio unit; and setting all components in the RX path in a power-on status; detecting a gain of the TX observation module by means of a signal from the TR switch passing through the TX observation module and a gain of the RX path by means of a TX leakage signal passing through the RX path; calculating a gain difference between the TX observation module and the RX path; and comparing the calculated gain difference with a preset gain difference to determine a state of the RX path. With the TDD radio unit and RX supervision method, the TX leakage signal is used, and thus no extra simulation source is need. Furthermore, a real time supervision is performed.

Abstract: A computer-implemented method to select a user profile is disclosed. The method may include obtaining, from a first device, a communication attempt at a user device. The method may further include obtaining first device data associated with the communication attempt and with the first device. The method may also include comparing the first device data with multiple data elements of each of multiple user profiles associated with the user device. The method may further include selecting a user profile of the multiple user profiles that includes the data element that matches the first device data in response to the first device data matching a data element of the multiple data elements. The method may also include loading an audio profile for the selected user profile on the user device.

Abstract: Provided is a method and system for audio quality enhancement. The audio quality enhancement method may include determining whether a software audio quality enhancement function is desired and/or required by analyzing a microphone input signal that is input to the electronic device and a speaker output signal that is output from the electronic device; and activating or inactivating the software audio quality enhancement function based on a result of determining whether the software audio quality enhancement function is desired and/or required.

Abstract: Context sensitive communication augmentation is disclosed. A user equipment (UE) can receive a trigger related to determining a context of a UE. The trigger can be manual, automatic, or remote. In response to receiving the trigger, the context of the UE can be determined based on sensor data. The sensor data can be received from a sensor of the UE or another sensor. Based on the context of the UE, a metric related to a communication modality of the UE can be determined. The metric can be communicated to another device. Information related to the context of the UE, including location information, can be communicated to the other device. A communication modality can be selected based on the metric relate to the UE and communication modalities of the other device. The metric can be updated based on changes in the context of the UE, allowing for updating the communication modality.

Abstract: A computing system, computer-implemented method, and computer program product are provided for facilitating behavior based notifications concerning multiparty communication threads. According to an implementation, the computer-implemented method may include defining a rules set associated with a user for participation in a multiparty communication thread. The method may also include evaluating a plurality of communications associated with the multiparty communication thread. The method may also include determining when a given communication associated with the multiparty communication thread satisfies one or more of the rules included within the rules set. The method may further include delivering the given communication based upon, at least in part, a determining that the given communication satisfies the one or more rules.

Abstract: A computer-implemented method to select a user profile is disclosed. The method may include obtaining, from a first device, a communication attempt at a user device. The method may further include obtaining first device data associated with the communication attempt and with the first device. The method may also include comparing the first device data with multiple data elements of each of multiple user profiles associated with the user device. The method may further include selecting a user profile of the multiple user profiles that includes the data element that matches the first device data in response to the first device data matching a data element of the multiple data elements. The method may also include loading an audio profile for the selected user profile on the user device.

Abstract: Methods, systems, and products provide remote notification of calls. When a call is received, a contact list is consulted for a contact. An electronic message is sent to an address of the contact. The electronic message notifies the contact of the call.

Abstract: A microphone assembly includes an acoustic sensor and a voice activity detector on an integrated circuit coupled to an external-device interface. The acoustic sensor produces an electrical signal representative of acoustic energy detected by the sensor. A filter bank separates data representative of the acoustic energy into a plurality of frequency bands. A power tracker obtains a power estimate for at least one band, including a first estimate based on relatively fast changes in a power metric of the data and a second estimate based on relatively slow changes in a power metric of the data. The presence of voice activity in the electrical signal is based upon the power estimate.

Abstract: Methods and systems for identifying sound from a source of interest are provided for herein. In some embodiments, a first audio feed is captured by a first microphone and a second audio feed is captured by a second microphone. The first microphone may be located closer in proximity to the source of interest than the second microphone. The first audio feed can be processed utilizing the second audio feed to produce a first processed audio feed that can enable identification of sound originating from the source of interest. In some embodiments, the second audio feed can be additionally processed utilizing the first audio feed to produce a second processed audio feed. In such embodiments, frequencies from the first processed audio feed can be compared against frequencies of the second processed audio feed to identify sound originating from the source of interest. Other embodiments may be described and/or claimed herein.

Abstract: An amplifier system is described. The amplifier system may amplify an audio signal transmitted via a connected loudspeaker in a first mode of operation. In a second mode of operation the amplifier system may amplify a signal generated by the loudspeaker operated in reverse as a microphone. Because a loudspeaker is less sensitive than a microphone the amplifier system may be used to acquire audio at high sound pressure levels for example at a concert or while making a phone-call in a noisy environment for example with a high level of wind noise.

Abstract: An earphone device, a portable terminal and an earphone system are provided. The earphone system includes a portable terminal having an earphone interface, an earphone device electrically connectable to the earphone interface, and at least one device recognition element which is disposed in at least one of the earphone interface and the earphone device and which provides a certain resistance value when the earphone device is electrically connected to the earphone interface so that the portable terminal determines that the earphone device is electrically connected to the earphone interface.

Abstract: Systems and methods are provided for echo reduction. For example, first audio data associated with a current time unit is obtained, the first audio data being generated by an audio-acquisition device of a receiver based on at least information associated with a first audio signal acquired by the audio-acquisition device and a second audio signal output from an audio-playback device of the receiver, the second audio signal being output from the audio-playback device of the receiver based on at least information associated with second audio data transmitted previously by a sender before the current time unit; third audio data transmitted by the sender during a second time unit preceding the current time unit is obtained; and fourth audio data is obtained by performing echo reduction on the first audio data based on at least information associated with the third audio data.

Abstract: Audio frames are classified as either speech, non-transient background noise, or transient noise events. Probabilities of speech or transient noise event, or other metrics may be calculated to indicate confidence in classification. Frames classified as speech or noise events are not used in updating models (e.g., spectral subtraction noise estimates, silence model, background energy estimates, signal-to-noise ratio) of non-transient background noise. Frame classification affects acceptance/rejection of recognition hypothesis. Classifications and other audio related information may be determined by circuitry in a headset, and sent (e.g., wirelessly) to a separate processor-based recognition device.

Abstract: Modern electronic devices are becoming smaller while the need to critically locate acoustic sensors or transducers, e.g., microphones, is increasing in order to provide improved intelligibility in adverse environments. The acoustic transducers can be used in arrays with their outputs processed in beam forming and noise cancellation algorithms. The use and creation of features on the surface of a small package to deal with the problem of achieving high performance solutions in a limited space are disclosed herein. For example, embodiments of electronic communication devices (10) include acoustic transducers (26) integrated with other features such as a volume control (34, 50), an antenna (87, 260, 280), an adapter (300) which can be coupled to an input/output jack (90) and an accessory connector, or a projection (98).

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 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: Disclosed are a method for implementing a video call with a Bluetooth-based headset and a video communication terminal. The method includes: looking up a Bluetooth headset; starting a VTCALL unit of the AP module and an AG unit; sending a VTCALL command and an AG command to the audio module; switching an audio channel to a BTVTCALL channel; starting a recording and playing mode of the video communication terminal, collecting uplink and downlink audio data, and transmitting the uplink and the downlink audio data to the Bluetooth module; transmitting the uplink and the downlink audio data to the Bluetooth headset; executing the data exchange with the Bluetooth headset for implementing a Bluetooth video call. The present invention implements the Bluetooth video call by modifying software, thereby using the current hardware resources and decreasing the cost.

Abstract: An induced loss value is retrieved from a configuration file of an MTA device. The loss value is provided to a signal generator that generates call progress/alerting signals and FSK signals in the MTA telephony device. To mitigate the imbalance between incoming voice signal levels and call progress/alerting signals that may occur because the call progress/alerting signals are not generated at a level equal to that of the incoming voice signals, the loss/attenuation amount from the configuration file is used to adjust the levels of the generated call progress/alerting and FSK signals by a corresponding amount before they are attenuated along with the incoming voice signals.

Abstract: An embodiment of an apparatus for computing control information for a suppression filter for filtering a second audio signal to suppress an echo based on a first audio signal includes a computer having a value determiner for determining at least one energy-related value for a band-pass signal of at least two temporally successive data blocks of at least one signal of a group of signals. The computer further includes a mean value determiner for determining at least one mean value of the at least one determined energy-related value for the band-pass signal. The computer further includes a modifier for modifying the at least one energy-related value for the band-pass signal on the basis of the determined mean value for the band-pass signal. The computer further includes a control information computer for computing the control information for the suppression filter on the basis of the at least one modified energy-related value.

Abstract: A method for automatically configuring an interface unit (112, 212, 912, 1012, 1112) for connecting a headset (110, 210, 910, 1110) and a telephone (100) by electrical connections (992, 993, 1092, 1093). The electrical connections (992, 993, 1092, 1093) between the interface unit (112, 212, 912, 1012, 1112) and the telephone (100) comprise two pairs of electrical paths having a number of distinct permutation settings. Each of the number of distinct permutation settings have a complementary setting together forming a number of distinct complementary setting pairs. The method comprises the steps of: a) receiving a test signal, b) for each distinct permutation setting generating an output signal in response to the test signal, c) determining a least preferred distinct permutation setting based on the output signals, d) selecting the complementary setting to the least preferred distinct permutation setting as a preferred setting.

Abstract: Acoustic echo control and noise suppression in telecommunication systems. The proposed method of processing multi-channels audio loudspeakers signals and at least one microphone signal, comprises the steps of: transforming the input microphone signals (y1 (n), y2 (n), . . . , yM (n)) into input microphone short-time spectra, computing a combined loudspeaker signal short-time spectrum [X(i,k)] from the loudspeaker signals, (x1 (n), x2 (n), . . . , xL (n)), computing a combined microphone signal short-time spectrum [Y(i,k)] from the input microphone signal, (y1 (n), y2 (n), . . . , yM (n)), estimating a magnitude or power spectrum of the echo in the combined microphone signal short-time spectrum, computing a gain filter (G(i,k)) for magnitude modification of the input microphone short-time spectra, applying the gain filter to at least one of the input microphone spectra, converting the filtered input microphone spectra into the time domain (e1 (n), e2 (n), . . . , eM (n)).

Abstract: A telephone includes a transmit channel and a receive channel, each including a bank of sub-band filters having a VAD coupled one to each sub-band filter. Each VAD measures the spectral energy in a sub-band, compares the spectral energy to a first threshold, and produces an output signal representative of whether or not the first threshold is exceeded. The voice activity detector also includes a threshold circuit for calculating a dynamically adjustable noise threshold based upon averaged measured spectral energy. A wide band or system VAD monitors echo canceling circuitry to detect voice activity and double talk.

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: 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, methods, and other embodiments associated with echo cancellation are described. According to one embodiment, a method uses two signals from a transmitter as input to a look-up table to generate a non-linear echo cancellation signal. The non-linear echo cancellation signal is applied to a receiving path to reduce non-linear echo distortion on the receiving path based at least in part on the current signal and the previous signal.

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 circuits for isolating and attenuating signals generated by a telephone network. In disclosed embodiments, a metering pulse signal is isolated from the terminals of the connecting device, and then attenuated with an impedance that is synthesized with a programmable digital signal processor. Embodiments also utilize the digital signal processor to synthesize a termination impedance for the connecting device. The termination impedance matches closely the characteristic impedance of the network, so as to minimize wave reflections and the like.

Abstract: The invention provides a single digital communication link between system-side and line-side circuitry in a DAA, capable both of carrying data signals and of transferring a substantial amount of power to the line-side circuitry. The invention comprises a system-side interface circuit, a line-side interface circuit, and an isolation barrier including a transformer. Each interface circuit is capable of connection to an upstream communication circuit (either line-side or system-side), from which it may receive data signals to be transmitted across the isolation barrier to the other interface circuit, and to which it may pass data signals received across the isolation barrier from the other interface circuit. The line-side interface circuit may further include a rectifier and a storage device.

Abstract: The present invention relates to an electronic device that can output received voice and input transmitted voice at the same time and enhances a suppression function for echo due to the received voice sneaking into the transmitted voice. An electronic device (e.g., portable telephone terminal device) outputs the received voice from a voice output unit (speaker), inputs the transmitted voice through a voice input unit (microphone), and includes an echo canceller unit that subtracts a pseudo echo signal for the received voice from the transmitted voice to suppress an echo component in the transmitted voice and a controlling unit that changes an echo suppression amount of the echo canceller unit in accordance with the received voice volume.

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: 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 method (200) for improving quality of output audio (126). The method can include detecting an output acoustic signal (128) and generating a receive audio signal (134) based, at least in part, on the detected output acoustic signal. A frequency domain representation (140) of the receive audio signal can be compared to a frequency domain representation (138) of a source audio signal (124) from which the output acoustic signal is generated. At least one distortion signal (142) in the receive audio signal can be identified, and the source audio signal can be selectively equalized to reduce an amplitude of the source audio signal at a frequency that correlates to the distortion signal.

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: The invention provides a single digital communication link between system-side and line-side circuitry in a DAA, capable both of carrying data signals and of transferring a substantial amount of power to the line-side circuitry. The invention comprises a system-side interface circuit, a line-side interface circuit, and an isolation barrier including a transformer. Each interface circuit is capable of connection to an upstream communication circuit (either line-side or system-side), from which it may receive data signals to be transmitted across the isolation barrier to the other interface circuit, and to which it may pass data signals received across the isolation barrier from the other interface circuit. The line-side interface circuit may further include a rectifier and a storage device.

Abstract: A method and an apparatus for Voice Activity Detection (VAD) and an encoder are provided. The method for VAD includes: acquiring a fluctuant feature value of a background noise when an input signal is the background noise, in which the fluctuant feature value is used to represent fluctuation of the background noise; performing adaptive adjustment on a VAD decision criterion related parameter according to the fluctuant feature value; and performing VAD decision on the input signal by using the decision criterion related parameter on which the adaptive adjustment is performed. The method, the apparatus, and the encoder can be adaptive to fluctuation of the background noise to perform VAD decision, so as to enhance the VAD decision performance, save limited channel bandwidth resources, and use the channel bandwidth efficiently.

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: The invention provides a single digital communication link between system-side and line-side circuitry in a DAA, capable both of carrying data signals and of transferring a substantial amount of power to the line-side circuitry. The invention comprises a system-side interface circuit, a line-side interface circuit, and an isolation barrier including a transformer. Each interface circuit is capable of connection to an upstream communication circuit (either line-side or system-side), from which it may receive data signals to be transmitted across the isolation barrier to the other interface circuit, and to which it may pass data signals received across the isolation barrier from the other interface circuit. The line-side interface circuit may further include a rectifier and a storage device.

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: In an on-board hands-free communication apparatus, a gain of an amplifier of a speaker for outputting an incoming voice signal from a telephone device varies with a volume set value of a speaker volume switch adjusted by an user. When the gain of the amplifier increases equal to or greater than a threshold value, a feature extraction amplification circuit performs a feature extraction amplification processing to apparently amplify output power of the incoming voice signal, and a control circuit fixes the gain of the amplifier to the threshold value regardless of the volume set value.

Abstract: Acoustic echo control and noise suppression is an important part of any “handsfree” telecommunication system, such as telephony or audio or video conferencing systems. Bandwidth and computational complexity constraints have prevented that stereo or multi-channel telecommunication systems have been widely applied. The advantages are very low complexity, high robustness, scalability to multi-channel audio without a need for loudspeaker signal distortion, and efficient integration of echo and noise control in the same algorithm.