Abstract: Systems, methods, and computer program products for frequency-domain estimation of latency between audio signals. In some embodiments, the estimation is performed on first blocks of data indicative of samples of a first audio signal and second blocks of data indicative of samples of a second audio signal, and includes determining a coarse latency estimate, including by determining gains which, when applied to some of the second blocks, determine estimates of one of the first blocks, and identifying one of the estimates as having a best spectral match to said one of the first blocks. A refined latency estimate is determined from the coarse estimate and some of the gains. Optionally, at least one metric indicative of confidence in the refined latency estimate is generated. Audio processing (e.g., echo cancellation) may be performed on the frequency-domain data, including by performing time alignment based on the refined latency estimate.

Abstract: An XTC circuit includes delay circuits, differentiated signal generating circuits, and an amplitude adjusting and adding circuit. A signal Da, which is one aggressor signal, is input to the differentiated signal generating circuit after being delayed by the delay circuit, and the differentiated signal generating circuit generates a differentiated signal having a differentiated waveform of the signal Da. In the amplitude adjusting and adding circuit, the differentiated signal generated by the differentiated signal generating circuit is amplitude-adjusted to become a current signal, and the differentiated signal after the amplitude adjustment is current-added to the signal Db.

Abstract: In a communication system, a timing-dependence cancelling module is included for cancelling timing-dependence of a transmission signal, so as to render a timing-dependent signal be capable of being utilized on communication systems. Besides, updating an echo cancelling parameter by applying an error difference variable and a data difference variable, or by directly decreasing a step-size coefficient, may also fulfill the purpose of reducing or eliminating timing dependence in a transmission signal of a communication system.

Abstract: A method includes receiving an ultrasound signal at an ultrasound receiver from an ultrasound transmitter. The method also includes detecting movement of at least one object based on the received ultrasound signal and at least one previously received ultrasound signal. The method further includes modifying a parameter of an acoustic echo canceller in response to the detected movement.

Abstract: Crosstalk effects can be taken into account in automatic test pattern generation (ATPG) by providing crosstalk fault models, determining paths and/or nodes to be sensitized to activate each crosstalk fault, and optimizing to enable as many crosstalk faults as possible with a given pattern, subject to constraints. Constraints can include threshold numbers of endpoints/observation points and/or attempts to sensitize. Intermediate nodes in a crosstalk fault model path to an observation point can also be determined and/or sensitized.

Abstract: A method includes receiving at least a first input audio channel and a second input audio channel, and using an inter-channel prediction model to form at least one inter-channel parameter. The first and second input audio channels represent a spatial audio image of an acoustic space. The inter-channel prediction model is a linear prediction model representing a predicted sample of the first input audio channel using a weighted linear combination of samples of the second input audio channel. An apparatus for practicing the method and a corresponding computer program product are also disclosed.

Abstract: An input signal is processed through noise suppression (NS) and echo control (EC) via a multipath model that reduces noise pumping effects while maintaining EC performance. A copy of a “noisy” input signal is sent to an EC component before the noisy signal is sent to a NS component, which processes the signal first, when there is a consistent noise level for estimation. The copy of the pre-processing noisy signal is sent to the EC component along with a “clean” or “noise-suppressed” signal output from the NS component. The EC component analyzes the noisy signal as if the EC was the first component in the signal chain to determine what actions to take. The EC component then applies these actions to the clean signal received from the NS component.

Abstract: Methods, apparatuses, and systems are presented for performing adaptive equalization involving receiving a signal originating from a channel associated with inter-symbol interference, filtering the signal using a filter having a plurality of adjustable tap weights to produce a filtered signal, and adaptively updating each of the plurality of adjustable tap weights to a new value to reduce effects of inter-symbol interference, wherein each of the plurality of adjustable tap weights is adaptively updated to take into account a constraint relating to a measure of error in the filtered signal and a constraint relating to group delay associated with the filter. Each of the plurality of adjustable tap weights may be adaptively updated to drive group delay associated with the filter toward a target group delay.

Abstract: A sound processing apparatus includes: a sound output unit which outputs sound based on a first sound signal as first sound processing when execution of the first sound processing is instructed; a sound collecting unit which generates an echo signal by collecting surrounding sound; a filtering unit which performs filtering process with an adaptive digital filter on the first sound signal and generates a pseudo echo signal; a subtracting unit which removes a sound component, which is output from the sound output unit and reaches the sound collecting unit, from the echo signal by subtracting the pseudo echo signal from the echo signal to generate a residual signal; an updating unit which updates the adaptive digital filter based on the residual signal and the first sound signal; and an update resolution control unit which controls the resolution at which the adaptive digital filter is updated by the updating unit.

Abstract: Test equipment including a signal generator and a method of generating a test signal associated with an overall frequency band are disclosed. Also disclosed is a method of using the test signal to test an echo canceller. The method of generating the test signal includes generating a first time segment associated with a first frequency sub-band of the overall frequency band and generating a second time segment associated with a second frequency sub-band of the overall frequency band. The second frequency sub-band is higher than the first frequency sub-band. The method further includes generating a time gap segment separating the first time segment and the second time segment.

Abstract: A method and apparatus for canceling an echo in audio communication is disclosed. The method comprises receiving an audio signal from a network and subsequently detecting a mixture audio signal comprising a target audio signal and an echo audio signal, the echo signal corresponding to the received audio signal. The method then comprises estimating the target audio signal by determining magnitude spectrograms for the mixture and received audio signals respectively, estimating a magnitude spectrogram of the target audio signal dependent on those of the mixture and received audio signal, and generating an output audio signal that estimates the target audio signal, the output audio signal being dependent on the estimated magnitude spectrogram.

Abstract: An input signal is supplied to a loudspeaker-room-microphone system having a transfer function and that provides an output signal. An adaptive filter unit models the transfer function of the loudspeaker-room-microphone system and provides an approximated output signal, where the output signal and the approximated output signal are subtracted from each other to provide an error signal.

Abstract: Various embodiments of a wireless multimode receiver having an off-chip duplex filter associated with a multimode band, and a blocker cancellation circuit disposed on a semiconductor chip are described in the present disclosure.

Abstract: A communication apparatus is disclosed including: an analog-front-end circuit for receiving and processing an analog input signal; an analog-to-digital converter (ADC) coupled with the analog-front-end circuit for converting processed signal from the analog-front-end circuit into a digital input signal; and a control unit coupled with the ADC for adjusting at least one resistance and/or at least one capacitance in an analog echo cancellation circuit according to the digital input signal before the analog-front-end circuit receives a training sequence that is the first training sequence transmitted from a second communication apparatus after the second communication apparatus begins communicating with the communication apparatus.

Abstract: A signal generator produces a victim signal having crosstalk emulation by filtering and combining a victim signal waveform record file and an aggressor signal waveform record file generated using parameters selected by a user. A signal channel or a cascaded signal channel is characterized using one or more S-parameter arrays. The S-parameter array or arrays represent a mixed-mode multiple-port device under test. Coefficients of a NEXT filter, a FEXT filter and a forward transmission filter are derived from selected S-parameters of the S-parameter array. The aggressor signal is filtered individually by the NEXT and FEXT filters. The victim signal is summed with the filtered aggressor signal from the NEXT filter with the resulting summed signal being filtered by the forward transmission filter. The filtered signal from the forward transmission filter is summed with the filtered aggressor signal from the FEXT filter to generate a victim signal having crosstalk emulation.

Abstract: An integrated hybrid circuit includes a transmission unit, a transceiver coil, a transceiver circuit, a hybrid matching circuit, and a receiving circuit. The transmission unit generates a pair of upstream signals according to a user transmission signal. The transceiver coil transmits the pair of upstream signals to a central office through a pair of twisted pair, and receiving a pair of downstream signals from the central office through the pair of twisted pair. The hybrid matching circuit receives an adjustment signal to adjust selective impedances. The receiving circuit receives the pair of downstream signals from the central office, and generates the adjustment signal according to downstream and upstream rates and signals from the hybrid matching circuit. The transmission unit adjusts transmission power and bandwidth of the transmission unit and the receiving unit adjusts filter bandwidth of the receiving unit according to the adjustment signal for optimizing upstream and downstream rates.

Abstract: A system and method for provide a stable gain from an adaptive gain control device in a signal path. An echo canceller is also located in the signal path, and is used to provide performance information regarding losses in the signal. This performance information is fed to the automatic gain control device via a connection. The automatic gain control device thereafter uses the performance information to determine a maximum gain that might be provided based upon losses cause by echo conditions. The gain however is limited in order to provide for a stable system. The performance information includes a loss rate that includes a combination of the echo return loss and the echo return loss enhancement.

Abstract: The present invention relates to design and implementation of low complexity adaptive echo and NEXT cancellers in multi-channel data transmission systems. In this invention, a highly efficient weight update scheme is proposed to reduce the computational cost of the weight update part in adaptive echo and NEXT cancellers. Based on the proposed scheme, the hardware complexity of the weight update part can be further reduced by applying the word-length reduction technique. The proposed scheme is general and suitable for real applications such as design of a low complexity transceiver in 10GBase-T. Different with prior work, this invention considers the complexity reduction in weight update part of the adaptive filters such that the overall complexity of these adaptive cancellers can be significantly reduced.

Abstract: Echo cancellation is handled using a pattern classification technique. During a training phase, the communications device is trained to learn patterns of signal inputs that correspond to certain communication modes. After numerous different patterns have been classified by mode, the classified patterns are used during real-time use of the communications device in order to determine, dynamically, the mode in which the communications device is currently operating. The communications device may then apply, to the microphone-produced signal, a suppression action.

Abstract: Methods and techniques to implement a digital signal processor for avoidance of audio feedback are disclosed, in particular, audio signal processing systems that reduce the requirement for physical segregation of sound acquisition and diffusion zones. In a more general aspect, the components and techniques described herein provide a for a sound space and sound processing equipment such that sound travelling electronically in a loop through the sound processing equipment that is output into a physical sound diffusion zone, received at the input to the sound processing equipment, and then re-amplified, etc. is attenuated over that loop by frequency modification. The frequency modification is such that, at least for some signals, on each pass through the loop, the sound processing equipment will attenuate or amplify individual sub-bands of the frequency spectrum of the audio signal that is received at the input of the sound processing equipment.

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: The coefficient generating section receives a first signal which is the output signal of the microphone of a signal generated by subtracting the output signal of a linear echo canceller from the output signal of the microphone and a second signal which is the output signal of the linear echo canceller. The coefficient generating section detects the minimum value of the variation with time of the ratio of the amplitude of the first signal to that of the second signal and outputs the value of constant times the detected minimum value as a crosstalk coefficient indicating the degree of crosstalk of the echo. The converting section corrects the first signal according to the crosstalk coefficient and the second signal to generate a near-end signal which is the resultant signal of when the echo is removed from the first signal and outputs the near-end signal to an output terminal.

Abstract: A howling suppression device that can reduce quality deterioration of processed sound includes: a delay unit delaying the input signal to output the delayed input signal as a reference signal; a signal separation unit including an adaptive filter extracting a periodic signal component from the reference signal by adaptively updating a filter coefficient; a howling detection unit detecting an occurrence of howling using at least a signal of the periodic signal component output from the adaptive filter; and a howling suppression unit. The howling suppression unit includes: a suppression filter obtaining the updated filter coefficient from the adaptive filter with timing when the howling detection unit detects the occurrence of the howling, to extract the periodic signal component from the reference signal based on the filter coefficient; and a subtractor subtracting the periodic signal component from the input signal so as to output a signal obtained by the subtraction.

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: Transmit equalization over high speed digital communication paths may be compensated in a receiver for a probe on that path. In one example, a probe input provides a signal from an electronic communications path, the signal having been processed by a transmit equalizer. A filter circuit processes the signal to compensate for the transmit equalizer, and a decoder decodes the processed signal and produces an output for use by test equipment.

Abstract: Disclosed methods and systems measure acoustic responses to training signals activated prior to communication sessions. Profiles related to the acoustic responses are saved and adapted during communication sessions. Training signals may have uniform frequency distributions over a frequency range and may be in response to user inputs, timeouts, or predetermined events. In the next excessive divergence is detected, an adapted profile may be substituted by an original, trained profile.

Abstract: Embodiments of a multi-channel audio communication device and methods for reducing echoes are generally described herein. Other embodiments may be described and claimed. In some embodiments, echo-reduction circuitry may insert training signals into digital audio signals at or below a noise floor and in a non-audible portion of the frequency spectrum based on a spectral mask. Training signals may be generated from spectrum estimates and training sequences. An adaptive filter may generate echo-cancellation signals using filter coefficients generated fawn the training sequences and return-path signals. The echo cancellation signals may remove echo signals from the return-path signals.

Abstract: An adaptive audio system can be implemented in a communication device. The adaptive audio system can enhance voice in an audio signal received by the communication device to increase intelligibility of the voice. The audio system can adapt the audio enhancement based at least in part on levels of environmental content, such as noise, that are received by the communication device. For higher levels of environmental content, for example, the audio system might apply the audio enhancement more aggressively. Additionally, the adaptive audio system can detect substantially periodic content in the environmental content. The adaptive audio system can further adapt the audio enhancement responsive to the environmental content.

Abstract: A communication end device of a two-way communication system is shown. The device includes an audio signal capture device for capturing local audio to be transmitted to another end device, an audio signal rendering device for playing remote audio received from the other end device, and buffers for buffering the captured and rendered audio signals. The device also includes an audio echo canceller operating to predict echo from the rendered audio signal at a calculated relative offset in the captured audio signal based on an adaptive filter, and subtract the predicted echo from the signal transmitted to the other end device The calculated relative offset that is used by the audio echo canceller for a current signal sample is adjusted if a difference between it and an adjusted relative offset of a preceding sample exceeds a threshold value.

Abstract: Systems and methods for a headset adapter circuit are disclosed. The headset adapter circuit generally includes a maximal length sequence generator circuit coupled to the adapter transmit signal line for injecting a maximal length sequence signal. A maximal length sequence cross-correlator circuit is coupled to the adapter receive signal line for capturing a receive signal sample and convolving the receive signal sample with the maximal length sequence signal to generate a host phone sidetone path impulse response.

Abstract: In one embodiment, an acoustic echo control (AEC) module receives an outgoing signal and an incoming signal, which, at various times, contains acoustic echo corresponding to the outgoing signal. The AEC module has a delay estimation block that estimates, in the time domain, the echo delay using an adaptive filtering technique. This delay estimation is used to align samples of the incoming signal having acoustic echo with the corresponding samples of the outgoing signal from which the acoustic echo originated. The AEC module determines whether or not samples of the incoming signal contain acoustic echo based on the aligned outgoing signal, and the determinations are applied to a hangover counter. The AEC module then suppresses acoustic echo in the incoming signal and adds comfort noise to the incoming signal. The amount of echo suppression performed is gradually increased or decreased based on comparisons of the counter to a hangover threshold.

Abstract: A sound pickup apparatus selecting one of a plurality of microphones and performing echo cancellation processing for a plurality of microphones with an echo canceller to output a sound. The sound pickup apparatus sets a “learning mode” when the power supply is turned on, outputs a calibration sound from an echo cancellation calibration sound generator via a speaker, detects an echo at that time with a microphone and obtains an echo cancellation use parameter canceling the echo.

Abstract: A method and apparatus for removing an echo signal in a signal transmission/reception apparatus of a communication system are provided. A signal transmission/reception apparatus determines an echo channel impulse response using a reception signal, generates an echo signal removing coefficient using the echo channel impulse response, removes an echo signal from the reception signal using the echo signal removing coefficient, and transmits a signal in which the echo signal is removed.

Abstract: A system includes microphone devices, echo cancellation mechanism, and speaker devices. Each echo cancellation mechanism corresponds to one of the microphone devices. Each speaker device corresponds to a sound source signal. Each sound source signal is spatially distributed to all the speaker devices, and each echo cancellation mechanism receives as a reference signal a combination of all the sound source signals.

Abstract: A communication apparatus is disclosed including: an analog-front-end circuit for receiving and processing an analog input signal; an analog-to-digital converter (ADC) coupled with the analog-front-end circuit for converting processed signal from the analog-front-end circuit into a digital input signal; and a control unit coupled with the ADC for adjusting at least one resistance and/or at least one capacitance in an analog echo cancellation circuit according to the digital input signal before the analog-front-end circuit receives a training sequence that is the first training sequence transmitted from a second communication apparatus after the second communication apparatus begins communicating with the communication apparatus.

Abstract: Embodiments of an audio communication device and methods for reducing echoes are generally described herein. Other embodiments may be described and claimed. In some embodiments, echo-reduction circuitry may insert a training signal into digital audio signals at or below a noise floor and in a non-audible portion of the frequency spectrum based on a spectral mask. The training signal may be generated from a spectrum estimate and a training sequence. An adaptive filter may generate an echo-cancellation signal using filter coefficients generated from the training sequence and return-path signals. The echo-cancellation signal may remove echo signals from the return-path signals.

Abstract: A sound processing apparatus includes: a sound output unit which outputs sound based on a first sound signal as first sound processing when execution of the first sound processing is instructed; a sound collecting unit which generates an echo signal by collecting surrounding sound; a filtering unit which performs filtering process with an adaptive digital filter on the first sound signal and generates a pseudo echo signal; a subtracting unit which removes a sound component, which is output from the sound output unit and reaches the sound collecting unit, from the echo signal by subtracting the pseudo echo signal from the echo signal to generate a residual signal; an updating unit which updates the adaptive digital filter based on the residual signal and the first sound signal; and an update resolution control unit which controls the resolution at which the adaptive digital filter is updated by the updating unit.

Abstract: In an echo processing method and device which can detect an accurate echo section without effects of a far end signal, an echo delay, and a reduction of an echo cancellation amount, a signal of a specified frequency band is generated in conformity with a near end signal, and the signal of the specified frequency band is added to the near end signal to form a transmitting signal. Receiving signals are separated into the signal of the specified frequency band and a signal of a band other than the specified frequency band. An echo section is detected based on the signal of the specified frequency band separated. An echo component in the signal of the band other than the specified frequency band is removed and a level of the echo component is detected based on the near end signal in the echo section.

Abstract: Testing equipment for and a method of performance testing an echo canceller are disclosed. The method includes generating a white noise test signal that includes a plurality of high-level white noise time segments, each of which is separated by a low-level white noise time gap. The generated white noise test signal is transmitted to the echo canceller. At least one echo output signal is received from the echo canceller, and the echo output signal results from the white noise test signal. The level and duration of the received at least one echo output signal are measured.

Abstract: A method and system for determining and compensating for a non-linearity in a hands-free acoustic telecommunication device is disclosed. The method determines a back electromotive force signal induced in a loudspeaker from at least one of a coil voltage, a current signal and estimates of coil resistance and inductance, estimating at least one of a cone position and a cone velocity from the BEMF integrated with respect to time and determining an estimate of an echo value from a series connection of an estimated inverse of a force factor function primitive and the estimated acoustic impulse response; and outputting the estimated echo value.

Abstract: A wireless repeater includes an internal feedback path for adaptively cancelling an echo between an output antenna and an input antenna. The internal feedback path employs an adaptive algorithm implementing a list having a plurality of list elements. Each list element has one or more echo cancellation parameters and one or more repeater settings. The list may be pruned by employing a minimum distance between elements within the list. A method for stabilizing a wireless repeater includes obtaining an autocorrelation of a signal in a signal path of the repeater, detecting an echo in the signal path based on the autocorrelation, providing a list of elements indicating past successful echo cancellation coefficients and associated repeater settings, and adaptively adjusting an error of the echo cancellation coefficients to cancel the echo in the signal path.

Abstract: In one embodiment, the line termination card includes a data output terminal configured to output a data sequence. The card further includes a vectoring entity configured to parse and encode the data sequence into frequency samples according to a carrier loading parameter, configured to scale the frequency samples into scaled frequency samples according to a carrier scaling parameter, and, configured to process the scaled frequency samples for crosstalk compensation. A controller is configured to adjust the carrier loading parameter and the carrier scaling parameter, and a forwarder is coupled to the data output terminal and to the controller. The forwarder is configured to forward the data sequence, the carrier loading parameter and the carrier scaling parameter towards a further line termination card.

Abstract: A method of determining if an echo canceller comprised in a communication channel that is used to transmit at least one tonal signal and characterized by a model echo path of the channel, is acceptably adapted to cancel echoes in the channel, the method comprising: determining a performance measure how well the canceller cancels echoes of at least one test signal characterized by at least one frequency different from a frequency of the at least one tonal signal; and determining if the canceller is acceptably adapted responsive to the measure.

Abstract: A method and apparatus for removing an echo signal in a signal transmission/reception apparatus of a communication system is provided. A signal transmission/reception apparatus estimates an input channel response using a training sequence, generates a first signal by removing the input channel response from a first reception signal, detects an echo channel impulse response using the first signal, detects an echo signal removing coefficient using the echo channel impulse response, generates a second signal in which an echo signal is removed by applying the echo signal removing coefficient to a second reception signal, and removes the second signal from a third signal, wherein the first signal is received prior to receiving the second signal and the second signal is received prior to receiving the third signal.

Abstract: The method comprises, in the time domain, echo cancellation processing (40) by subtracting from the microphone signal a reference signal delivered by an adaptive circuit for modeling the acoustic coupling between the microphone (18) and the loudspeaker (16). Successive frames of the signal are subjected in the frequency domain to processing (42) for suppressing the residual echo that remains after the echo cancellation, together with noise reduction processing (44). The residual echo suppression processing comprises estimating respective values for the power of the residual echo over a plurality of defined frequency bands for successive signal frames, then calculating and applying a variable gain specific to each frequency band and to each frame, the respective gains being smaller whenever the estimated relative power of the residual echo is high for the frequency band under consideration, and vice versa.

Abstract: A signal correction device including: an orthogonal transform section configured to perform an orthogonal transform for an input signal that includes a speech as a target signal and an unnecessary non-target signal; an interval determining section configured to determine whether each frame of the input signal is an interval in which the non-target signal is dominantly included; a suppressing gain calculating section configured to calculate suppressing gain for suppressing the non-target signal for each first frequency band width for a frame determined to be the interval, and to calculate suppressing gain for suppressing the non-target signal for each second frequency bandwidth for a frame determined not to be the interval; and a signal correcting section configured to perform a signal correcting process for suppressing the non-target signal for a transform coefficient acquired by the orthogonal transform section by using the suppressing gain.

Abstract: A feedback calibration system and a method for controlling an electronic signal are disclosed. The feedback calibration system includes an input controller adapted to modify an input signal in response to a control signal and generate a modified input signal, a signal processing block including a signal analyzer, wherein the signal processing block is adapted to process the modified input signal to generate an output signal and the signal analyzer is adapted to detect an undesirable condition of the output signal and transmit a detection signal corresponding to the undesirable condition, a transfer function estimator adapted to model and transmit a transfer function estimate of the signal processing block in real-time in response to the detection signal, and a programmable device adapted to transmit the control signal to the input controller for modifying the input signal, wherein the control signal is based upon the transfer function estimate.

Abstract: A method for synchronizing audio signals of a communication device includes determining whether a downlink thread or a uplink thread is running, pausing the downlink thread or the uplink thread, and executing the downlink thread or the uplink thread by directing a scheduler. A system and storage medium with instructions for performance of the system also provided.

Abstract: A communication device is provided that includes a speaker for outputting a ringing signal, a microphone for receiving the ringing signal, and logic configured to analyze the spectral content of the received ringing signal and adjust noise reduction parameters and echo cancellation parameters based on the analyzed spectral content of the received ringing signal.

Abstract: A transmit signal second-order inter-modulation (IM2) canceller for a portable handset using a full duplex mode of operation (e.g., WCDMA) is used to controllably reduce IM2 introduced by a transmit signal that appears in a received signal in a receive channel of the portable handset. The transmit signal IM2 canceller includes a delay estimator and a digital signal adjuster. The delay estimator receives a first input from a receive channel and a second input from a transmit channel. The delay estimator generates an estimate of the IM2 that the transmit channel introduces in the receive channel. The digital signal adjuster removes the estimate of the IM2 before forwarding a modified receive channel signal to a baseband subsystem of the portable handset.