Abstract: Technologies are generally described for a system for measuring a quality of experience (QoE). In some examples, a quality of experience (QoE) measuring device may include a background noise detecting unit configured to measure a background noise around a near-end talker, and a decision unit configured to determine whether a double talk event detected by a double talk detector is caused by the background noise around the near-end talker.

Abstract: A radio frequency (RF) transmitter is provided. The RF transmitter includes first and second drivers that are configured to receive first and second sets of complementary RF signals. Restoration circuits are coupled to the first and second drivers, and a bridge circuit is coupled to the first and second restoration circuits. By having the restoration circuits and the bridge circuit, a common mode impedance and a differential impedance can be provided, where the common mode impedance is lower than the differential impedance.

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

Abstract: A method and apparatus for managing audio conflicts and priorities is disclosed. A mobile device (110) initiates a navigation application (410) and an audio application (415), each configured to utilize audio input and/or output components. The mobile device identifies (430, 435) a priority of a communication associated with the audio application and manages (440, 445) the audio input and output components based on the priority and other factors. In particular, the mobile device or components thereof can delay (440) an indication of the communication until a sufficient break in a route traversal is reached, can mute (445) the audio input component while the navigation application utilizes the audio output component, and/or can perform other functions related to the audio conflicts and priorities of the applications.

Abstract: A system for echo cancellation includes a dynamic speaker management (DSM) module, a current/voltage sensing amplifier, a sound pressure level (SPL) model module and an echo canceller. The example DSM module is receptive to a far-end signal and is operative to develop a modified far-end signal and a plurality of parameter outputs. The example current/voltage sensing amplifier is coupled to the modified far-end signal and develops an amplifier output, a voltage (V) parameter output, and a current (I) parameter output. The example sound pressure level (SPL) model module is coupled to the plurality of parameter outputs of the of the DSM module and is operative to develop a predicted SPL. The example echo canceller module is responsive to the predicted SPL and to a near-end signal and operative to develop an echo-canceled output signal.

Abstract: A method includes accessing signal data descriptive of a transmission sequence and pre-determined values associated with the transmission sequence. The signal data and the pre-determined values may be stored in a memory. The method includes transmitting a signal from a speaker of an electronic device according to the transmission sequence. The method includes generating a frame based on one or more signals received at a microphone of the electronic device. The one or more signals include an echo signal associated with the transmitted signal. The method includes processing the frame using the pre-determined values to produce an output frame in which a contribution associated with the echo signal is reduced.

Abstract: A mobile station is capable of detecting an insertion of a headset without a separate detection pin and removing noise generated when the headset is separated. A method includes detecting an insertion of a headset by a mobile station. An apparatus for detecting an insertion of an headset into a mobile station includes an headset attachment/detachment detecting circuit unit for transmitting an insertion signal or a separation signal to a controller by using a resistance value of the headset attached/detached to/from the mobile station; and a controller for, when an headset separation signal is received from the headset attachment/detachment detecting circuit unit, controlling such that an output of noise to the headset, which is not completely separated from the mobile station, is blocked by turning off a microphone bias.

Abstract: According to an aspect, a mobile communication device includes a housing, a speaker, a microphone, a detecting unit, and a processing unit. The speaker is provided in the housing, and outputs an incoming voice according to an incoming voice signal. The microphone is provided in the housing. The microphone receives an outgoing voice and outputs an outgoing voice signal in response to reception of the outgoing voice. The detecting unit detects vibration of the housing and outputs a housing-vibration signal indicating the vibration of the housing. The processing unit performs echo cancellation to the outgoing voice signal based on the incoming voice signal and the housing-vibration signal.

Abstract: The present disclosure relates to the technical field of mobile terminals, and discloses a mobile terminal and a volume adjusting method and device thereof. The volume adjusting method comprises the following steps of: acquiring a noise pressure signal from surroundings of the mobile terminal; comparing a volume value corresponding to the acquired noise pressure signal with a volume control value currently set in the mobile terminal; adjusting the volume control value of the mobile terminal according to the comparison result; and controlling a volume level of the mobile terminal according to the volume control value that has been adjusted.

Abstract: The present invention relates to a method of suppressing noise in a communication device. The idea consists in forwarding to a noise suppression module some information regarding the radio transmission in terms of radio activity and/or in terms of radio transmission power. The module then advantageously uses this information to suppress radio path noise.

Abstract: A noise cancellation system for an audio system such as a mobile phone handset, or a wireless phone headset has a first input for receiving a first audio signal from one or more microphone positioned to receive ambient noise, and a second input for receiving a second audio signal from a microphone positioned to detect the user's speech, as well as a third input for receiving a third audio signal for example representing the speech of a person to whom the user is talking. A first noise cancellation block receives the first audio signal and generates a first noise cancellation signal, and this is combined with the third audio signal to form a first audio output signal. A second noise cancellation block receives at least a part of the first audio signal and said second audio signal and applying noise cancellation to generate a second audio output signal.

Abstract: An device includes an audio output unit, a first audio input unit, a second audio input unit that are disposed in a position closer to the audio output unit than the first audio input unit, a unit for outputting a combined signal of which audio signals from the first and second audio input unit are combined so as forming directivity in which sensitivity in a direction of the audio output unit is low when viewed from the first and second audio input units, a unit for generating artificial echo corresponding to an echo component mixed in the audio inputted to the first audio input unit, and a unit for performing an echo suppression process to the combined signal by using the artificial echo.

Abstract: A method for reducing a disturbance in an input signal caused by an output signal in a multi-port connector, a multi-port connector circuit for reducing a disturbance in an input signal caused by an output signal, and a mobile device are described.

Abstract: The invention relates to a closed loop control system for active noise reduction, comprising a speaker and an adding device connected to the speaker. A feedforward control and a feedback control each comprise a microphone for recording noise interference or for recording a sound output by the speaker. Control networks for forming a corresponding control parameter are coupled to the corresponding microphones and are connected to the adding device at the output side thereof According to the invention, the feedback control for noise reduction is adapted on the basis of a first acoustic ratio and the feedforward control for noise reduction is adapted on the basis of a second acoustic ratio.; The control network of the feedback control is designed to at least partially compensate for the control parameter of the feedforward control when current acoustic ratios change in the direction of the first acoustic ratio.

Abstract: Residual frequency components of a reference signal are suppressed from an error signal. A magnitude of the frequency domain representation of the reference signal is divided by a magnitude of the frequency domain representation of LMS-filtered representation of the error signal to obtain a frequency domain ratio of the frequency domain representation of the reference signal to the frequency domain representation of the LMS-filtered signal. The frequency domain ratio of the frequency domain representation of the reference signal to the frequency domain representation of the LMS-filtered signal is multiplied by the frequency domain ratio of the frequency domain representation of the reference signal to the frequency domain representation of the LMS-filtered signal, to obtain a frequency domain signal having reduced residual frequency components of the reference signal.

Abstract: A transceiver mitigates signal leakage into a receive path from a transmit path. A subtraction circuit determines a difference between a receive signal and a compensation signal to produce a compensated receive signal prior to demodulation by a demodulator. An equalizer both amplitude adjusts and phase adjusts orthogonal baseband transmit signals based on the difference from the subtraction circuit to produce the compensation signal. A digital tuning circuit determines at least one amplitude adjust coefficient to be used by the equalizer. The equalizer can have a polarity switch or a variable attenuator or a variable delay.

Abstract: A communication device includes memory, an input interface, a processing module, and a transmitter. The processing module receives a digital signal from the input interface, wherein the digital signal includes a desired digital signal component and an undesired digital signal component. The processing module identifies one of a plurality of codebooks based on the undesired digital signal component. The processing module then identifies a codebook entry from the one of the plurality of codebooks based on the desired digital signal component to produce a selected codebook entry. The processing module then generates a coded signal based on the selected codebook entry, wherein the coded signal includes a substantially unattenuated representation of the desired digital signal component and an attenuated representation of the undesired digital signal component. The transmitter converts the coded signal into an outbound signal in accordance with a signaling protocol and transmits it.

Abstract: A communication device including multiple microphones is provided. The communication device includes at least two microphones. The communication device further includes a sidetone feedback notifier for producing a notification signal. The sidetone feedback notifier is coupled to the microphones. The notification signal is based on the combination of, a first input audio signal provided for by a first microphone, and a second input audio signal provided for by a second microphone. The sidetone feedback notifier is coupled to a notification device for providing a feedback signal to a user based on the notification signal.

Abstract: A noise suppressing device receives sound signals through a plurality of sound-receiving units and suppresses noise components included in the input sound signals. The noise suppressing device includes a detecting unit which detects a usage pattern of the noise suppressing device from a plurality of usage patterns in which positional relationships of the plurality of sound-receiving units and/or positional relationships between the plurality of sound-receiving units and a target sound source are different from each other, a converting unit which converts using environment information used in a noise suppressing process to each of the sound signals inputted by the plurality of sound-receiving units into using environment information in accordance with a usage pattern detected by the detecting unit and a suppressing unit which performs the noise suppressing process using the using environment information converted by the converting unit to the sound signals.

Abstract: Residual frequency components of a reference signal are suppressed from an error signal. A magnitude of the frequency domain representation of the reference signal is divided by a magnitude of the frequency domain representation of LMS-filtered representation of the error signal to obtain a frequency domain ratio of the frequency domain representation of the reference signal to the frequency domain representation of the LMS-filtered signal. The frequency domain ratio of the frequency domain representation of the reference signal to the frequency domain representation of the LMS-filtered signal is multiplied by the frequency domain ratio of the frequency domain representation of the reference signal to the frequency domain representation of the LMS-filtered signal, to obtain a frequency domain signal having reduced residual frequency components of the reference signal.

Abstract: A method in a communications device includes the following operations. During a call, a process automatically detects that the device has moved from an at-the-ear position to an away-from-the-ear position. Based on the detection, a noise suppressor that operates upon an uplink signal for the call is signaled to change its noise suppression performance. Other embodiments are also described and claimed.

Abstract: An echo canceller for improved recognition and removal of an echo from a communication device. The echo canceller can dynamically reduce echo using an improved energy estimator and an improved adaptive filter. The improved energy estimator can determine if conversation is in a single talk period or a double talk period based on the combined energy of both the near end background noise and speech. The improved adaptive filter can reduce echo by dynamically changing adaptation speed or step size. In double talk, the adaptive filter(s) can dynamically slow-down or stop adaptation. In single talk, the filter can dynamically increase the speed of adaptation to improve accuracy, or decrease adaptation speed for stability.

Abstract: A sensor network system including node devices connected in a network via predetermined propagation paths collects data measured at each node device to be aggregated into one base station via a time-synchronized sensor network system. The base station calculates a position of the signal source based on the angle estimation value of the signal from each node device and position information thereof, designates a node device located nearest to the signal source as a cluster head node device, and transmits information of the position of the signal source and the designated cluster head node device to each node device, to cluster each node device located within the number of hops from the cluster head node device as a node device belonging to each cluster. Each node device performs an emphasizing process on the received signal from the signal source, and transmits an emphasized signal to the base station.

Abstract: A system and methods are provided whereby a user of a mobile device can adaptively switch the mobile device from speakerphone mode to handset mode without needing to look at the mobile device or without the need to activate a button or key sequence. In a preferred embodiment, modules are provided in the mobile device which allow for the detection of a voice conversation, the sampling of the user's voice and the switching between the two modes of operation. If the user's voice volume is above a certain threshold the mobile device operates in handset mode. If the user's voice volume is below a certain threshold the mobile device operates in handset mode. Through the inclusion of the embodiments described herein, a mobile device can allow a user to safely and quickly switch from one mode of operation to the other without a requirement for additional hardware in the mobile device.

Abstract: A method of removing a signal from among received signals, the method including: filtering the received signals; detecting a time band of the filtered received signals where an energy value of the filtered received signals exceeds a reference energy value; and applying a gain value to one or more received signals, from among the received signals, in the detected time band.

Abstract: Example methods, apparatus and articles of manufacture to detect echo during teleconferences are disclosed. A disclosed example method includes collecting first signal data for a first teleconference leg of a teleconference, collecting second signal data for a second teleconference leg of the teleconference, computing an echo detection metric from the first and second signal data, and comparing the echo detection metric to a threshold to determine whether an echo is likely present on the first teleconference leg of the teleconference.

Abstract: A transmitter includes a Power Amplifier (PA), an antenna, at least one passive component and control circuitry. The PA is controlled by a PA control voltage, is operative to amplify a Radio Frequency (RF) signal and has input and output amplifier terminals. The passive component has an input component terminal coupled to the output amplifier terminal of the PA and an output component terminal coupled to the antenna. The control circuitry is configured to determine an interim power level at the output amplifier terminal that causes the signal at the output component terminal to have a target output power level, to determine, based on the interim power level, a given PA control voltage that makes the interim power level producible by the PA, so that the signal at the output component terminal has the target output power level, and to apply the given PA control voltage to the PA.

Abstract: One or more embodiments describe controlling audio signals at a user device during a communication session between the user device and a remote node, in which a primary audio signal is received at audio input means of the user device for transmission to the remote node in the communication session. It is determined whether the user device is operating in a first or a second mode. In dependence upon determining that the user device is operating in the first mode, the secondary audio signals are selectively suppressed from being output from the user device during the communication session.

Abstract: A wireless headset has improved immunity to RF electromagnetic interference produced from wireless communications devices. A headset body is adapted to be worn by a user and includes a microphone and earpiece. An antenna receives wireless communication signals and passes them to RF and audio circuitry mounted within the headset body. The RF and audio circuitry include a Bluetooth module operatively connected to the antenna for transmitting and receiving wireless communication signals, an audio CODEC connected to the Bluetooth module, and audio connection lines connected between the CODEC and the earpiece and between the CODEC and the microphone. A filter is connected into each of the audio connection lines at the earpiece and microphone and operative for reducing the RF coupling from a mobile wireless communications device.

Abstract: Active noise cancellation (ANC) circuitry is coupled to the input of an earpiece speaker in a portable audio device, to control the ambient acoustic noise outside of the device and that may be heard by a user of the device. A microphone is to pickup sound emitted from the earpiece speaker, as well as the ambient acoustic noise. Control circuitry deactivates the ANC in response to determining that an estimate of how much sound emitted from the earpiece speaker has been corrupted by noise indicates insufficient corruption by noise. In another embodiment, the ANC decision is in response to determining that an estimate of the ambient noise level is greater than a threshold level of an audio artifact that could be induced by the ANC. Other embodiments are also described and claimed.

Abstract: A data transmission circuit transmitting an activation signal prior to a data signal through a signal transmission line, including: an activation detection signal generation unit for generating an activation detection signal by detecting the activation signal; and a wakeup signal generation unit for being activated by the activation detection signal, and generating a wakeup signal by detecting that the activation signal is transmitted for a predetermined time.

Abstract: An audio signal decorrelator for deriving an output audio signal from an input audio signal has a frequency analyzer for extracting from the input audio signal a first partial signal descriptive of an audio content in a first audio frequency range and a second partial signal descriptive of an audio content in a second audio frequency range having higher frequencies compared to the second audio frequency range. A partial signal modifier modifies the first and second partial signals, to obtain first and second processed partial signals, so that a modulation amplitude of a time variant phase shift or time variant delay applied to the first partial signal is higher than that applied to the second partial signal, or for modifying only the first partial signal. A signal combiner combines the first and second processed partial signals, or combines the first processed partial signal and the second partial signal, to obtain an output audio signal.

Abstract: A system and method for localized noise cancellation. An audio signal is received from an environment in close proximity to a primary area. The audio signal is processed to generate an inverse signal of the audio signal. The inverse signal is broadcast within the primary area to destructively interfere with the audio signal. The inverse signal is configured to prevent the audio signal from being broadcast through a telephone conversation ongoing in the primary area.

Abstract: An echo cancellation circuit in a full duplex two-way communication system comprising: an input/output terminal; a subtractor having a positive and a negative input terminals, in which a first transmission signal is inputted to the negative input terminal as a pseudo echo signal, the first transmission signal is inputted through an output buffer to the positive input terminal as an echo signal, the pseudo echo signal inputted to the negative input terminal is subtracted from the echo signal inputted to the positive input terminal; and a result of the subtraction is outputted; and an echo cancellation error reducing unit having a D/A converter at an input side or an output side of the subtractor.

Abstract: An adaptive filter configured to use multiple algorithm species that differ in the quality of echo suppression and respective burdens imposed on the computational resources of the host communication device. Depending on the available computational budget, the adaptive filter selects an algorithm species that, while supporting a relatively high quality of echo suppression, involves a relatively low risk of overwhelming the computational resources. The adaptive filter monitors changes in the available computational budget and, if appropriate or necessary, can change the algorithm species to maintain a quality of echo suppression that is optimal for the current computational budget. If a change of the algorithm species is initiated, then at least a portion of internal algorithm data from the previously running algorithm species might be transferred for use in the subsequent algorithm species.

Abstract: This method comprises the following steps in the frequency domain: a) estimating a probability that speech is present; b) estimating a spectral covariance matrix of the noise picked up by the sensors, this estimation being modulated by the probability that speech is present; c) estimating the transfer functions of the acoustic channels between the source of speech and at least some of the sensors relative to a reference constituted by the signal picked up by one of the sensors, this estimation being modulated by the probability that speech is present; d) calculating an optimal linear projector giving a single combined signal from the signals picked up by at least some of the sensors, from the spectral covariance matrix, and from the estimated transfer functions; and e) on the basis of the probability that speech is present and of the combined signal output from the projector, selectively reducing the noise by applying variable gain.

Abstract: An audio interface adapted to reduce a subscriber voice may receive a subscriber voice and a background noise. The subscriber voice may then be compared to the to the background noise. If the received subscriber voice is louder than the received background noise, the audio interface may output a message to the cellular telephone subscriber indicating the subscriber may reduce his speaking volume. Additionally, an audio interface may process a voice waveform that corresponds to the subscriber voice and a background waveform that corresponds to the background noise to generate a substantially opposite voice waveform and a substantially opposite background waveform respectively. The substantially opposite voice waveform and background waveform may be substantially out of phase from the voice waveform and background waveform respectively and output via one or more output ports of the audio interface.

Abstract: An echo canceler circuit (10) and method attenuates at least post-echo canceler uplink data (90) to produce attenuated uplink data (100) in response to uplink echo return loss based attenuation data (40). The echo canceler circuit (10) includes an echo return loss based attenuation data generator (20) and at least an uplink data attenuator (30). The echo return loss based attenuation data generator (20) produces the uplink echo return loss based attenuation data (40) in response to echo return loss data (70). The echo return loss data (70) is based on at least one of: attenuated downlink data (50), pre-echo canceler uplink data (60), and/or amplifier gain data (80). The uplink data attenuator (30) attenuates the post-echo canceler uplink data (90) to produce attenuated uplink data (100) based on the uplink echo return loss based attenuation data (40).

Abstract: This includes systems and methods for noise cancellation and power management in a wireless headset. The wireless headset can perform noise cancellation by using two or more omni-directional microphones to generate a noise canceling “cone.” Audio signals received outside of the cone can be filtered out. An accelerometer can be used to determine the location of the wireless headset and the system can then properly position the cone to face the user's voice and filter out other audio signals. Additionally, by monitoring the ambient noise, the system can save power by turning off the noise cancellation system when the amount of ambient noise is below a threshold value.

Abstract: A printed circuit board for a wireless communications device includes an acoustic path, and acoustically couples microphone circuitry mounted to a surface of the printed circuit board to an exterior of the wireless communications device. A portion of the acoustic path is integrally formed with the printed circuit board and extends generally parallel to the surface of the printed circuit board. One end of the acoustic path is disposed proximate the microphone circuitry and an opposing end opens to receive acoustic sound from the exterior of the wireless communications device. Audible sound propagates through the acoustic path to the microphone circuitry.

Abstract: A vehicle communication system detects the presence of a passenger wearable communication device. The system receives audio signals from multiple sources inside or outside of a vehicle. The system processes the signals before routing the signals to multiple destinations. The destinations may include wearable personal communication devices, front and/or rear speakers, and/or a remote mobile device.

Abstract: A multi-function communications device has a processor that generates a user interface audible tone signal. The device also has a downlink digital signal processor, and an uplink digital signal processor. A mixer has an input to receive the downlink signal and another input to receive the user interface tone signal. The uplink processor has an acoustic echo canceller having an input to receive the uplink signal and another input to receive an output from the mixer. The echo canceller may reduce the amount of both the far-end user's speech and the user interface tone that may be present in the uplink signal. The mixer may be positioned within the chain of audio signal processors, or it may be positioned outside the chain. Other embodiments are also described and claimed.

Abstract: A multimedia conferencing system includes a loud speaker system, one or more microphones for receiving a local audio signal and a remote audio signal, a state machine and an echo canceller that operates in conjunction with two reference signals to remove substantially all of a feedback signal component in the local audio signal that results from reinforcing and playing the local audio signal over the loud speaker system. The state machine operates to detect that only the local audio is active, and if so controls the operation of the echo canceller such that only the feedback component of the local audio signal is removed and the local audio signal is not suppressed.

Abstract: Techniques to characterize and reduce acoustic echo are described. For example, a mobile computing device may comprise an audio characterization module to send a first signal to a first component and receive a second signal from a second component. The first signal may be a deterministic audio training sequence and the second signal may be a recovered audio training sequence. The audio characterization module compares the first signal and the second signal to determine an audio echo path. Other embodiments are described and claimed.

Abstract: A wireless conference phone system has a base station to couple at least one conference unit to a telephone network, such as the public switched telephone network (PSTN) or a digital telephone network such as a voice over internet protocol (VoIP) network. Each conference unit performs echo cancellation of the audio signal received from a remote location allowing a simplified base station implementation.

Abstract: A receiver suppresses noise simultaneously using a noise suppressor and an automatic gain controller (AGC). The receiver divides a received signal into an audio period and a non-audio period, analyzes whether the noise characteristic of the non-audio period corresponds to a non-static noise or a static noise, and if the noise characteristic corresponds to the static noise, analyzes whether the static noise is caused by a network or a transmitter side terminal. In accordance with the analyzed noise characteristic, the noise suppressor firstly suppresses the noise by determining the noise suppressing intensity, and sends a signal, from which the noise has been firstly suppressed, to the AGC. The AGC secondly suppresses the noise included in the signal. In this case, the threshold value of the AGC is controlled in real time in accordance with the noise characteristic.

Abstract: A method for performing a call between a near-end user and a far-end user, which includes the following operations performed during the call by the near-end user's communications device. Automatic gain control (AGC) is performed to update a gain applied to an uplink speech signal. A frame is detected in a downlink signal that contains speech; in response, the updating of the gain is frozen. Other embodiments are also described and claimed.

Abstract: A speakerphone system includes first and second speakerphones connected via a network and a first processor in the first speakerphone and a second processor in the second speakerphone, each of the speakerphones having a first, active, mode for participating in a call with a telephone and a second, idle, mode, wherein, when the first speakerphone is operating in the first mode and the second speakerphone is operating in the second mode, the first speakerphone is adapted to send voice data packets to the second speakerphone, the second speakerphone is adapted to receive voice data packets from the first speakerphone, perform echo cancellation processing on the voice data packets and send the processed voice data packets back to the first speakerphone, and the first speakerphone is adapted to receive the processed voice data packets from the second speakerphone and use the processed voice data packets in the call.

Abstract: One aspect of the present invention is directed to an apparatus to perform impulse blanking of a received signal at multiple locations of a signal processing path. To effect such impulse blanking, multiple impulse detectors and blankers may be present, in addition to other circuitry. The impulse detectors may operate at different bandwidths, and the impulse blankers may be located at different locations of the signal processing path and may be differently configured.

Abstract: A communications system network that enables secondary use of spectrum on a non-interference basis is disclosed. Each secondary transceiver measures the background spectrum. The system uses a modulation method to measure the background signals that eliminates self-generated interference and also identifies the secondary signal to all primary users via on/off amplitude modulation, allowing easy resolution of interference claims. The system uses high-processing gain probe waveforms that enable propagation measurements to be made with minimal interference to the primary users. The system measures background signals and identifies the types of nearby receivers and modifies the local frequency assignments to minimize interference caused by a secondary system due to non-linear mixing interference and interference caused by out-of-band transmitted signals (phase noise, harmonics, and spurs).