Abstract: An outdoor radio communication system comprises a first radio unit, a second radio unit, and a single cable coupling the first radio unit to the second radio unit. Each radio unit includes a downconverter, a radio processor that is communicatively coupled to the downconverter, and a XPIC module. The cable further includes a first twisted-pair of wires for communicatively coupling the first downconverter to the second XPIC module and a second twisted-pair of wires for communicatively coupling the second downconverter to the first XPIC module. The first XPIC module generates a first reference signal using a signal from the second downconverter to cancel cross-polarization interference in an output signal of the first radio processor. Similarly, the second XPIC module generates a second reference signal using a signal from the first downconverter to cancel cross-polarization interference in an output signal of the second radio processor.

Abstract: A mobile terminal including a transceiver having a plurality of output terminals; a plurality of power amplifier units respectively connected to the plurality of output terminals, and configured to amplify a first or second signal output by the output terminals and to output the amplified first or second signal to a plurality of antennas respectively connected the plurality of power amplifiers; and a receiving module configured to receive a third signal through a receiving antenna disposed to be spaced apart from the plurality of antennas and to output the received third signal to the transceiver. Further, the receiving module includes first and second filters configured to control an intermodulation distortion (IMD) signal included in the third signal due to the output of the first or second signal.

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

Abstract: There is described a time-to-digital conversion scheme using an arrangement of delay elements based Time-to-Digital Converter, TDC (20), wherein dithering is built in the digital domain and introduced in the analog domain as a modulation of a supply voltage (TDC-supply) supplying delay elements of the TDC, each having a propagation delay which exhibits a dependency to their supply voltage.

Abstract: A controller for the conference session receives at least one audio signal to generate a speaker signal. The controller correlates the speaker signal with network timing information and generates speaker timing information. The controller transmits the correlated speaker signal and timing information to a mobile device participating in the conference session. The mobile device generates an echo cancelled microphone signal from a microphone of the mobile device, and transmits the echo cancelled signal back to the controller. The controller also receives array microphone signals associated with an array of microphones at known positions in the room. The controller removes acoustic echo from the array microphone signals, and estimates a relative location of the mobile device. The controller dynamically selects as audio output corresponding to the mobile device location either (a) the array microphone signal, or (b) the echo cancelled microphone signal from the mobile device.

Abstract: Improvements in voice signals transmitted within communication systems are obtained by use of adaptive filters, front and rear microphones, noise cancelling systems and other means and methods. Disclosed embodiments include the use of directional microphones, primary inputs, secondary inputs, adaptive weight generators, canceller outputs to improve signal to noise ratios and other communication attributes.

Abstract: A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to measure the output of the transducer in order to control the adaptation of the anti-noise signal and to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit that performs the adaptive noise canceling (ANC) function also either adjusts the frequency response of the anti-noise signal with respect to the reference microphone signal, and/or by adjusting the response of the adaptive filter independent of the adaptation provided by the reference microphone signal.

Abstract: A voice communication end device performs quality checks to determine whether acoustic echo cancellation would be ineffective, such as due to noise or clock drift or discontinuities between incoming and outgoing voice channels. In the case where echo cancellation would prove ineffective, the device falls back on a tri-state voice switching operation that includes a bi-direction state in which both channels are on in full duplex operation, which provides a smoother transition switching between active channels. The tri-state voice switching supports both voluntary transitions where the active user voluntarily stops to yield the active channel, and forced transitions where the active user is forcedly interrupted by the other user speaking more loudly.

Abstract: Methods and devices for improved echo cancellation in high noise environments, such as in-car speakerphones and the like, are provided that improve cancellation convergence in a high noise environment and in the presence of double talk. A new update method enables a robust echo canceller to update with reduced concern for the canceller being in a receive state or a double talk state. If a signal generated by a local talker is substantially the same as high noise, in terms of corruption of the echo canceller, the echo canceller converges in a robust manner even with continuous high noise at the input.

Abstract: The application relates to a communication device, e.g. a speakerphone, comprising a microphone signal path, MSP, and a loudspeaker signal path, SSP, the microphone signal path comprising a microphone unit, an MSP-filter, and a transmitter unit operationally connected to each other and configured to transmit a processed signal originating from an input sound picked up by the microphone, the loudspeaker signal path comprising a receiver unit, an SSP-filter, and a loudspeaker unit operationally connected to each other and configured to provide an acoustic sound signal originating from a signal received by the receiver unit. The communication device comprises a control unit for dynamically controlling the filtering characteristics of the MSP and SSP-filters based on one or more control input signals. This has the advantage of providing a simple and flexible scheme for decreasing echo in a communication device, while ensuring an acceptable sound quality in the transmitted signal.

Abstract: An echo path of the echo in a received audio signal is modelled using an adaptive model to determine an adaptive model estimate of the echo. The adaptive model estimate is used to determine an estimate of the echo power of the echo in the received audio signal. The power of the received audio signal is determined. The estimate of the echo power and the determined power of the received audio signal are used to determine echo suppression gains. Temporal smoothing is applied to one or more of the echo suppression gains and the one or more smoothed echo suppression gains are used to apply echo suppression to the received audio signal, thereby suppressing the echo in the received audio signal, wherein the amount of smoothing applied to the echo suppression gains is varied according to a non-decreasing function of the frequency of the received audio signal.

Abstract: Echo removal techniques are described. As part of the echo removal, a first model estimate of the echo in the received audio signal is determined using a first model and a second model estimate of the echo is determined using a second model. A first accuracy value of the first model is determined according to a model accuracy measure, and a second accuracy value of the second model is determined according to the model accuracy measure. It is then determined if the first model is more accurate than the second model based on a comparison of the first accuracy value and the second accuracy value and the second model is selectively updated based on said comparison. Echo removal is applied to the received audio signal using only the second model estimate of the echo.

Abstract: Echo removal techniques are described. An echo path of the echo in a received audio signal is modelled using a first model to determine a first model estimate of the echo. The first model estimate is used to determine a first performance value according to a performance metric. The first performance value is compared with a threshold value. It is determined if the echo path can be deemed linear based on the comparison. If so, the first model estimate of the echo is used to remove the echo in the received audio signal. Otherwise, the echo path of the echo is switched to be modelled using a second model to determine a second model estimate of the echo, and the second model estimate of the echo is used to remove the echo in the received audio signal.

Abstract: A system performs residual echo suppression on a microphone signal that receives, e.g., voice commands, and that is exposed to echo from multiple speakers. An example is a smartphone that receives voice commands while the smartphone is playing music through stereo speakers. The system estimates residual echo level in different ways, and determines which estimate to use. The technique responds well to the difficult to handle scenario of a spatially quiescent image suddenly transitioning to a spatially rich image. Even in the face of such difficult scenarios, the system detects and removes residual echo from the microphone signal, instead of allowing the undesired residual echo to pass through.

Abstract: Echo removal techniques are described. As part of the echo removal, an adaptive model estimate of the echo in a received audio signal is determined using an adaptive model based on an outputted audio signal and the received audio signal. The adaptive model executes an algorithm comprising a convergence parameter to determine filter coefficients and uses said filter coefficients to filter the outputted audio signal to determine the adaptive model estimate of the echo. An accuracy value of the adaptive model is determined according to an echo return loss enhancement metric. The convergence parameter is updated based on the accuracy value. The adaptive model estimate of the echo is used to remove the echo in the received audio signal.

Abstract: A system for actively reducing noise at a listening point, includes an earphone housing, a transmitting transducer, a receiving transducer and a controller. The transmitting transducer converts a first electric signal into a first acoustic signal, and radiates the first acoustic signal along a first acoustic path having a first transfer characteristic and along a second acoustic path having a second transfer characteristic. The receiving transducer converts the first acoustic signal and ambient noise into a second electrical signal. The controller compensates for the ambient noise by providing a noise reducing electrical signal to the transmitting transducer. The noise reducing electrical signal is derived from a filtered electrical signal that is provided by filtering the second electrical signal with a third transfer characteristic. The second and the third transfer characteristics together model the first transfer characteristic.

Abstract: Methods, systems, and apparatus for controlling audio signal processing are described herein. In accordance with certain embodiments, a measure of coupling between at least one microphone and one speaker associated with an audio device is obtained. The measure of coupling may be used to control audio signal processing, such as sidetone generation and application of multi-channel decorrelation to the audio signal.

Abstract: Disclosed is a self-correlation receiver of a transceiver in which a transmitter shares an antenna with a receiver. The self-correlation receiver includes: a clamper which receives a received signal, limits the magnitude of the received signal within a predetermined range and outputs a clamping signal; and a mixer which receives the received signal and the clamping signal and outputs a signal having a difference frequency between the received signal and the clamping signal.

Abstract: Provided are a method for estimating a spectrum density of diffused noises. Also provided is a processor for implementing the method. The processor includes at least two sound receiving units and a spectrum density estimating unit for estimating spectrum density.

Abstract: Method of enabling a remote communications device (106) with a telematics functionality module (150) can include providing a docking apparatus (114) coupled to interface with a vehicle (109). The remote communications device is communicatively coupled to the docking apparatus, wherein the remote communications device is non-enabled with a telematics functionality module (150). The docking apparatus and the remote communications device enable the remote communications device with the telematics functionality module.

Abstract: A telephonic device having a speakerphone function has a loudspeaker and a plurality of microphones. The plurality of microphones are coupled to a plurality of beamformers which produce a first beamform having a spatial null in the direction of the loudspeaker and a second beamform having a spatial null in the direction of near-end signals. The two beamforms are then processed to facilitate echo reduction. By comparing the beamform powers, a state of double-talk can be determined and the determination can be used to enhance echo reduction associated with speakerphone functionality.

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 echo canceller and an echo cancellation method are provided. The echo canceller includes a self-adaptive filter, a voice signal detection portion and a path change detection portion; a far-end voice signal is propagated in an echo path through a speaker and is picked up by a microphone to form an echo signal. The self-adaptive filter is configured to receive the far-end voice signal as a training signal to simulate the echo path, and cancel the echo signal in a near-end signal. The voice signal detection portion is configured to detect a communication status, control the self-adaptive filter according to the communication status, and control startup of the path change detection portion according to the communication status. The path change detection portion is configured to detect whether a change occurs on the echo path, and control the self-adaptive filter according to whether the change occurs on the echo path.

Abstract: In an embodiment, an apparatus includes a determiner, converter, adapter, and modifier. The determiner is configured to generate a representation of a difference between a first frequency at which a first signal is sampled and a second frequency at which a second signal is sampled, and the converter is configured to generate a second sample of the first signal at a second time in response to the representation and a first sample of the first signal at a first time. The adapter is configured to generate a sample of a modifier signal in response to the second sample of the first signal, and the modifier is configured to generate a modified sample of the second signal in response to a sample of the second signal and the sample of the modifier signal. For example, such an apparatus may be able to reduce the magnitude of an echo signal in a system having an audio pickup (e.g., a microphone) near an audio output (e.g., a speaker).

Abstract: The present invention provides for echo cancellation circuitry and variable rate encoding circuitry to cooperate with one another to effectively provide comfort noise in an effective and efficient manner. The echo cancellation circuitry will use far-end signals to generate estimated echo signals, which correspond to the actual echo signals appearing in near-end signals. The estimated echo signals are essentially subtracted from the near-end signals in an effort to remove the actual echo signals from the near-end signals. The echo cancellation circuitry will monitor any residual echo signals in the resulting processed near-end signals and provide residual echo control signals that are indicative of whether the residual echo signals should be replaced with comfort noise. The residual echo control signals are used at least in part by the variable rate encoding circuitry to determine the encoding rate to use for encoding different portions of the processed near-end signals.

Abstract: A method for computing a gain control metric used in controlling gain in a wireless repeater operates to store correlation and normalization values associated with the gain control metric for the previous N samples in registers. For each new sample of the gain control input signal, the correlation and normalization values are computed by discarding the multiplication terms of the obsolete sample and adding the multiplication terms of the new sample to the stored correlation and normalization values. In this manner, the complexity of the computation is greatly reduced and the complexity of the computation does not increase with the integration length.

Abstract: An apparatus and method for cross clock domain interference cancellation is provided to a communication system which includes a transmitter operated in a first clock domain and a receiver operated in a second clock domain. The apparatus comprises a First-In-First-Out (FIFO) circuit and a cancellation signal generator. The FIFO circuit receives a digital transmission signal of the transmitter in the first clock domain, and outputs the digital transmission signal in the second clock domain according to an accumulated timing difference between the first and second clock domains. The cancellation signal generator generates a cancellation signal for canceling an interference signal received by the receiver according to the digital transmission signal outputted by the FIFO circuit. The interference signal is generated in response to the digital transmission signal.

Abstract: A system includes an active noise-reducing headset and an alert processor. The active noise reducing headset includes a communication module that establishes a wireless communication link, a microphone that generates a microphone signal in response to ambient noise, a speaker, a noise reduction circuit that generates a noise reduction signal in response to the microphone signal, and a control circuit that receives an incoming alert signal from the communication module and plays the alert signal over the speaker. The alert processor includes a second communication module that establishes the wireless link with the active noise-reducing headset, and a controller that receive an incoming alert signal, generates an alert notification in response to the alert signal, and transmits the alert notification to the active noise-reducing headset.

Abstract: An echo cancellation apparatus is connectable to a speaker configured to output speaker signals and a microphone configured to receive a sound from the speaker and including a plurality of microphone elements. The echo cancellation apparatus includes: a generating unit configured to generate a plurality of sensitivity signals having different sensitivity patterns which represent directionality of the microphone, based on a plurality of microphone signals obtained from the respective microphone signals; a delay estimating unit configured to determine a shortest delay time as an estimated delay time, the shortest delay time being a shortest one of delay times between the speaker signals and the microphone signals, the delay times being obtained from the respective sensitivity signals; and an echo suppressing unit configured to suppress echoes of the plurality of microphone signals using the estimated delay time.

Abstract: A hearing aid compatible mobile handset is provided with a parasitic element which is electrically connected to a groundplane of a circuit board. The parasitic element may have a length of around one-quarter wavelength of a frequency of interest. The parasitic element may be are arranged as a pair of parasitic elements. The parasitic element can be disposed in a location where the near field strength is near a peak so as to help reduce the near field level to a point that will allow operation of the handset adjacent a hearing aid.

Abstract: A method includes cycling through a plurality of microphone and speaker combinations in a mobile device in response to the mobile device being placed in speakerphone mode. The mobile device includes a plurality of microphones and at least one speaker. The method obtains acoustic echo data from an echo canceller for each microphone and speaker combination, calculates an acoustic isolation value for each combination and then selects a microphone and speaker combination based on the acoustic isolation value. The method may also include determining an echo spectrum for each microphone and speaker combination using the obtained acoustic echo data, and select an echo control profile based on a characteristic of the echo spectrum. The echo control profile is selected to further improve acoustic isolation for the selected microphone and speaker combination.

Abstract: A wireless conference call telephone system uses body-worn wired or wireless audio endpoints comprising microphone arrays and, optionally, speakers. These audio-endpoints, which include headsets, pendants, and clip-on microphones to name a few, are used to capture the user's voice and the resulting data may be used to remove echo and environmental acoustic noise. Each audio-endpoint transmits its audio to the telephony gateway, where noise and echo suppression can take place if not already performed on the audio-endpoint, and where each audio-endpoint's output can be labeled, integrated with the output of other audio-endpoints, and transmitted over one or more telephony channels of a telephone network. The noise and echo suppression can also be done on the audio-endpoint. The labeling of each user's output can be used by the outside caller's phone to spatially locate each user in space, increasing intelligibility.

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: A system that incorporates teachings of the present disclosure may include, for example, a communication device having a controller to monitor ambient noise in proximity to the communication device, monitor a transmitted voice signal associated with the communication device, adjust a receiver volume for the transmitted voice signal based at least in part on a comparison of the transmitted voice signal with the ambient noise, monitor a received voice signal associated with the communication device, and adjust a speaker volume for the received voice signal based at least in part on a comparison of the received voice signal with the ambient noise, where the adjusting of the receiver volume is independent of the adjusting of the speaker volume. Other embodiments are disclosed.

Abstract: A noise cancellation device (NCD) comprises a microphone, an analog-to-digital converter, a digital-to-analog converter, a rechargeable battery, and a processor. The NCD acquires an audio input, from an external device such as a stethoscope or a cell phone, and passes the analog data into an ADC (analog-to-digital converter) for signal conversion. The digitized signals are then passed to the processor for further processing. The processor contains all the processing functions such as preprocessing (divide the input data into frames and apply shaping function to each frame), short term Fourier transform (STFT), adaptive filtering, inverse STFT, and signal synthesis.

Abstract: In order to reduce interference in an audio signal during a call on a mobile communication device, a plurality of transforms of the audio signal is performed, each transform containing phase information and amplitude information of corresponding samples of the audio signal. The results of the transforms are then averaged in order to generating a compensation signal that can be subtracted from the audio signal.

Abstract: Coefficient generator generates a crosstalk coefficient that is a predetermined value and that is used to calculate the amount of crosstalk of an echo. Converter uses either the output signal of a sound pickup device or the signal obtained by subtracting the output signal of an echo canceller from the output signal of the sound pickup device as a first signal, corrects the first signal based on the crosstalk coefficient generated in coefficient generator, and produces a near-end signal obtained by removing the echo from the first signal.

Abstract: Disclosed herein, among other things, are apparatus and methods for wireless hearing assistance devices and in particular to a controllable range control for wireless hearing assistance device systems. An advertisement is used to allow the receiver to quickly receive streaming information, thus conserving power. In various embodiments, a system for communications with a hearing assistance device includes a wireless streaming device having a range control. The range control is configured to provide a first range for an advertisement transmission and a second range for a streaming information transmission. The first range is less than the second range, in various embodiments.

Abstract: The present invention relates to a signal processing system, comprising a number of microphones and loudspeakers, a hands-free set configured to receive a telephone signal from a remote party and to transmit a microphone signal supplied by at least one of the microphones to the remote party; an in-vehicle communication system configured to receive a microphone signal supplied by at least one of the microphones; receive the telephone signal; amplify the microphone signal to obtain at least one first output signal; output the at least one first output signal and/or a second output signal corresponding to the telephone signal to at least one of the loudspeakers; and wherein the signal processing systems is configured to detect speech activity in the telephone signal and to control the in-vehicle communication system to reduce amplification of the microphone signal by a damping factor, if it is detected that speech activity is present in the telephone signal.

Abstract: Some embodiments provide a method in a wireless device for reporting channel state information, CSI, for a CSI process. The CSI process corresponds to a reference signal resource and an associated interference measurement resource, IMR. According to the method, the wireless device obtains an interference power adjustment value. The wireless device estimates interference and noise based on the IMR, and on the interference power adjustment value. Furthermore, the wireless device determines channel state information based on an estimated effective channel measured based on the reference signal resource, and on the estimated interference and noise. Finally, the wireless device transmits the channel state information to a network node.

Abstract: The present invention discloses a method for processing a call signal of a mobile phone, the method includes detecting a current call state of the mobile phone; acquiring an environmental noise signal when the mobile phone enters an earphone call state; and performing a noise reduction processing to a call signal according to the environmental noise signal for acquiring a call signal with the noise reduction processing. The present invention further discloses a mobile phone. The present invention is capable of ensuring that a user is not affected by environmental noises for guaranteeing a better quality of a call when the user makes the call with an earphone.

Abstract: A signal transceiving method, applied to a signal transceiver, includes: adjusting to approximate a value of a clock frequency of a signal to be transmitted from the signal transceiver to a value of a clock frequency of a received signal; performing an echo cancellation operation; computing a distance between a first certification code transmitted by the signal transceiver and a second certification code received by the signal transceiver; and stopping the echo cancellation operation when the distance is smaller than a threshold value.

Abstract: A system and method for configuring a vehicle user interface that provides hands-free use of a mobile communications device by a vehicle operator. First, the mobile communications device located at a vehicle is linked with a vehicle user interface in the vehicle so that the driver or other occupant can carry out hands-free telephone calls through the mobile device. Then, an evaluation phone call is placed using the mobile device so that volume levels, noise, etc. can be checked. Based on an analysis of that evaluation call, at least one audio parameter of the vehicle user interface is configured and is thereafter used to improve the audio communication between the user and the vehicle user interface.

Abstract: A radio base station for use in a Wideband Code Division Multiple Access network. The radio base station comprises a receiver for receiving radio signals transmitted over the air by users, and a first processor for processing the received signals in order to recover signals for a first group of users, the “cancellers”, and a first set of control signals for a second group of users, the “cancellees”. The radio base station further comprises an interference canceller for performing interference cancellation on the received signals using the recovered signals in order to generate an interference cancelled signal, a second processor for subsequently processing the interference cancelled signal in order to recover a second set of control signals for the cancellees, and a controller for using the control signals to control the transmission of data towards said cancellees.

Abstract: Noise reduction is provided to audio captured by a headset by employing spatially separated microphones provided by a headset and a mobile phone. Primary audio is captured by a headset microphone and includes both voice audio and ambient noise. Secondary audio is captured by a mobile phone microphone and includes ambient audio. Noise reduction is performed using the primary and secondary audio to generate a noise-reduce audio.

Abstract: Although the duplexer in a full-duplex transceiver circuit may only be guaranteed by the duplexer manufacturer to have a transmit band rejection from its TX port to its RX port of a certain amount, and may only be guaranteed to have a receive band rejection of another amount, the actual transmit band rejection and the actual receive band rejection of a particular instance of the duplexer may be better than specified. Rather than consuming excess power in the receiver and/or transmitter in order to meet performance requirements assuming the duplexer only performs as well as specified, the duplexer's in-circuit performance is measured as part of a transmitter-to-receiver isolation determination. Transmitter and/or receiver power settings are reduced where possible to take advantage of the measured better-than-specified in-circuit duplexer performance, while still meeting transceiver performance requirements. Power settings are not changed during normal transmit and receive mode operation.

Abstract: A noise canceling headphone includes a microphone provided in a headphone housing, collecting external sound, a canceling signal generating circuit generating a canceling signal in response to a signal output from the microphone which cancels external sound passing through the housing without being attenuated by a sound insulating characteristic of the headphone, a compensating signal generating circuit generating a compensating signal that compensates for the external sound attenuated by the sound insulating characteristic of the headphone, an adding circuit that adds a musical signal input from the exterior to the compensating signal, and a speaker unit that outputs an output signal from the adding circuit and the canceling signal.

Abstract: The present invention relates to a method as well as to a computing device (20) for editing a noise-database (13) containing noise information, said noise information being derived from noise signals within an audio stream (19). In order to enhance possibilities to create and utilize context information which emerge from tracking noise signals from an audio stream, for example a telephone call, the above method is characterized by the following steps: A) in a localizing step (14), determining geographical data of the location the noise signals origin from; B) in an analyzing step (15), analyzing the noise signals with reference to the noise content; C) in a linking step, linking the analyzed noise signals to said geographical data to create noise information; D) in a storing step, storing said noise information within said noise-database (13).

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

Abstract: A mobile communication device is provided. The mobile communication device includes a terminal body, and a speaker module located in the terminal body. The speaker module includes an enclosure located in the terminal body, the enclosure defining a chamber therein, and a speaker having a front surface and a rear surface, the speaker being located in the enclosure such that the front surface of the speaker is exposed from the enclosure and the rear surface thereof is positioned within the chamber, to allow a sound generated from the rear surface to resonate within the chamber. A mutual interference between the sound generated from the front side of the sound emission part and the sound emitted from the rear side thereof can be reduced to thereby improve the performance of the middle and low sound.