Abstract: A method and apparatus for estimating speech intelligibility in a mobile communications network component handling two-way communication between two ends of a signal path. Test signals adapted for speech intelligibility measurements are inserted into the signal path to simulate two-way communication. Double-talk is detected during the communication, and speech intelligibility measurements are performed only during periods of double-talk. This enables the effect of echo to be taken into account while avoiding undesirable effects from non-linear processing, and comfort noise if present, in the signal path. Voice enhancement devices may then be adjusted in response to the estimated speech intelligibility.

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

Abstract: An echo canceller system includes a first echo canceller having a first voltage divider and an adaptable second voltage divider that is configured to generate a first replica of an echo. A second echo canceller is configured to generate a second replica of an echo and has tap values that are generated in response to an error signal. A controller is coupled to the first and second echo cancellers and includes a selection algorithm that responds to the tap values of the second echo canceller and selects a voltage divider value for the adaptable second voltage divider.

Abstract: A method of determining a number of operating points for spectrum management of a plurality of digital subscriber lines in a communications network, the method comprising the steps of: determining optimization criteria (Oi) for the digital subscriber lines; determining crosstalk gains (Gi) for the digital subscriber lines; determining an initial population (Pi) of operating points (x1, . . . , xZ) where each of the operating points comprises importance ratings for each digital subscriber line of the plurality of digital subscriber lines; calculating new values of crosstalk and power spectrum densities (S) for the digital subscriber lines, by executing spectrum balancing for each operating point until the optimization criteria are fulfilled; calculating a new population of operating points, by executing an evolutionary algorithm; and iteratively performing the steps of calculating until a predefined stopping criterion is fulfilled.

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: An adaptive analog echo/near-end crosstalk (NEXT) cancellation system includes a processor configured to receive a first digital signal corresponding to a first signal received by the system, receive a second digital signal corresponding to a second signal transmitted by the system, and generate a first error control signal based on each of the first digital signal and the second digital signal. A cancellation device is configured to generate a second error control signal based on a first analog signal corresponding to the first signal. The cancellation device selectively outputs the second error control signal or the first error control signal based on whether the system is operating in a first mode or a second mode, respectively. The cancellation device selectively outputs a cancellation signal according to one of the second error control signal and the first error control signal, and a second analog signal corresponding to the second signal.

Abstract: Clock synchronization for an acoustic echo canceller (AEC) with a speaker and a microphone connected over a digital link may be provided. A clock difference may be estimated by analyzing the speaker signal and the microphone signal in the digital domain. The clock synchronization may be combined in both hardware and software. This synchronization may be performed in two stages, first with coarse synchronization in hardware, then fine synchronization in software with, for example, a re-sampler.

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: A new acoustic echo suppressor and method for acoustic echo suppression is described herein. Exemplary embodiments of the acoustic echo suppressor use one linear regression model for each subband. The linear regression model for each subband may operate on the squared magnitude of the input samples as well as corresponding cross-products. In this way, accurate and robust estimates of the echo signal in each subband can be obtained, thereby providing good echo reduction while keeping the signal distortion low.

Abstract: A method is provided for encoding multiple microphone signals into a composite source-separable audio (SSA) signal, conducive for transmission over a voice network. The embodiments enable the processing of source separation of the target voice signal from its ambient sound to be performed at any point in the voice communication network, including the internet cloud. A multiplicity of processing is possible over the SSA signal, based on the intended voice application. The level of processing is adapted with the availability of the processing power at the chosen processing node in the network in one embodiment. An apparatus for separating out the target source voice from its ambient sound is also provided. The apparatus includes a directed source separation (DSS) unit, which processes the two virtual microphone signals in the SSA representation, to generate a new SSA signal including the enhanced target voice and the enhanced ambient noise.

Abstract: In a telephone device, a near-end signal processor delays each input near-end signal delivered from the respective microphones by a delay amount, and synthesizes the input near-end signals into a synthesized near-end signal. In an echo canceller, mounted on board the telephone device, a far-end signal is delayed by the same delay amounts as used in the near-end signal processor to form plural delayed far-end signals, which are synthesized into a synthesized far-end signal, with which the echo component in the synthesized near-end signal will be suppressed.

Abstract: An echo removing apparatus that can remove an echo appropriately if the produced sound volume becomes large although the configuration is simple is provided. A control section 17 switches the operations of an echo suppressor 14 and an echo canceller 15 in response to the volume value input from a volume operation section 18. If the volume value is large, the echo suppressor 14 is operated and an echo is suppressed reliably. If the volume value is small, the echo canceller 15 is operated and double talk is enabled.

Abstract: An echo cancelling algorithm in a communication device initializes a step size value used in an adaptive echo filter based on a background noise signal power level relative to a power level of a received signal and a power level of an echo estimate relative to an output of an echo canceller. The algorithm then adjusts the step size value. One aspect adjusts the step size based on the detection of large fast fourier transform values at one, or more, disturbing-signal frequencies. Another aspect estimates residual echo energy to adjust an estimated echo energy, which then is used to set a double talk flag if a transmit signal has much more power than the estimated echo signal. Another aspect compares transmit signal power to a decimated version of the transmit signal power and sets the double talk flag if the former exceeds the latter by a predetermined amount.

Abstract: A packet voice transceiver adapted to reside at a first end of a communication network and to send an ingress communication signal comprising voice packets to, and receive an egress communication signal comprising voice packets from, a second packet voice transceiver residing at a second end of the communication network. The packet voice transceiver includes a far-end echo canceller that reduces echo that is present in the egress communication signal. The far-end communicates with other functional components of the transceiver system and cancels echo or refrains from canceling echo based on the activity of the other functional components.

Abstract: Methods and systems for echo modulation are described. In one embodiment, intensities of a plurality of values in multiple windows of an audio signal may be obtained. The windows may be subject to a periodic boundary condition. A plurality of echo values may be calculated for each of the respective windows. The audio signal may be altered in one or more of the windows using a windowing function and echo values. Additional methods and systems are disclosed.

Abstract: A scalable decoder device (50) for signals representing audio comprises a primary decoder (21) connected to an input (40). The primary decoder (21) is arranged to provide a primary decoded signal (23) based on received parameters (4). A primary postfilter (31) is connected to the primary decoder (23) to provide a primary postfiltered signal (32). A secondary enhancement decoder (45) is connected to the input (40) and arranged to provide a secondary decoded enhancement signal (44). The device further comprises a combiner arrangement (55), arranged for combining the primary postfiltered signal (32) and a signal (53) based on the secondary decoded enhancement signal (44) into an output signal (6) to be provided at an output (6). The combining is made with an adaptable strength relation between contributions from the two signals. A method for decoding coded signals representing audio operates in analogy with the scalable decoder device (50).

Abstract: In an embodiment, the access node includes first and second transceiver units for initializing first and second communication channels, respectively, over first and second communication lines, respectively. The initialization of the first and second communication channels respectively including first and second handshake phases wherein peer transceiver units exchange capabilities and select (i) a common mode of operation, (ii) first and second crosstalk channel acquisition phases from the first and second communication lines, and (iii) first and second generic initialization phases. The access node further includes a line initialization controller configured to stretch the execution of the first handshake phase with respect to the execution of the second handshake phase so as to shorten a time difference between respective entries into the first and second crosstalk channel acquisition phases.

Abstract: A method and arrangement for cancelling echoes in a communication terminal (400) during a voice call with an opposite party. The communication terminal (400) comprises an echo cancelling unit (416,600) adapted to produce an estimate of echo information in received speech signals coming from a microphone in the communication terminal (400), and subtract the estimate from the received microphone signals, before being transmitted to the opposite party. The estimate is produced based on down-sampled received microphone signals and down-sampled received speech signals from the opposite party. The down-sampled speech signals from the opposite party are filtered in a digital filter (606), and the output signals from the filter (606) are up-sampled to form the estimate.

Abstract: A method and a device for data processing in a digital subscriber line environment, wherein a flat noise margin is determined based on an actual noise determined and based on a maximum noise expected. A device is initialized using the flat noise margin. Also, a communication system is described that includes such a device.

Abstract: Disclosed is an apparatus and method for cancelling a wideband acoustic echo that controls a data flow of data that is transmitted and received using a central processing unit, and calculates data using a calculation processing unit configured to be distinct from the central processing unit.

Abstract: Systems and methods for reducing crosstalk are provided. In one implementation, a method is provided. The method includes applying a pilot signal to a disturber twisted pair and measuring near end crosstalk of a victim twisted pair. The coupling characteristics between the disturber and the victim twisted pair are determined. A far end crosstalk function is derived for estimating the far end crosstalk in the victim twisted pair using the determined coupling characteristics. A crosstalk reduction signal is determined using the estimated far end cross talk. The crosstalk reduction signal is injected into the victim twisted pair to reduce crosstalk in the victim twisted pair.

Abstract: In an audio system having a microphone, a speaker coupled to a source of audio output, and an echo canceller coupled to the speaker and microphone, latency between the source of audio output and the speaker may be compensated in echo cancellation performed by the echo canceller. The echo canceller may use a reference signal derived from a signal from the source of audio output in echo cancellation. The latency may be compensated by measuring the latency between the signal from the source of audio output and the speaker, determining a delay amount from the latency, delaying the reference signal by the delay amount to produce a delayed reference signal, and using the delayed reference signal as the reference signal in the echo canceller.

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 system, apparatus, method, and computer-readable medium for coded-domain echo cancellation. The method includes receiving a signal including at least one packet, and replacing the at least one packet with a replacement packet. In one example, the replacement packet is a comfort noise packet (such as a SID_UPDATE packet) or a NO_DATA packet. In an example embodiment, the at least one packet included in the signal includes one or more comfort noise packets, and, prior to the replacing, the one or more comfort noise packet(s) are stored in a buffer. In another example, prior to the replacing, the at least one packet is compared to a reference packet to determine whether the at least one packet is an echo packet. The packet, in one example, is encoded based on an adaptive multi-rate (AMR) (e.g., AMR-NB or AMR-WB) codec.

Abstract: A filter coefficient setting device for setting a filter coefficient of an echo prevention device including a first FIR filter, and a second FIR filter, comprises: a filter coefficient initial setting portion configured to set a predetermined filter coefficient for the first and second FIR filters when the echo prevention device is started.

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: A preferred embodiment of an apparatus for computing filter coefficients for an adaptive filter for filtering a microphone signal so as to suppress an echo due to a loudspeaker signal includes an extractor for extracting a stationary component signal or a non-stationary component signal from the loudspeaker signal or from a signal derived from the loudspeaker signal, and a computer for computing the filter coefficients for the adaptive filter on the basis of the extracted stationary component signal or the extracted non-stationary component signal.

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 circuit includes a microphone circuit, an audio processing module, a digital audio processing module, and an active noise reduction (ANR) circuit. The microphone circuit receives acoustic vibrations and generates an audio signal therefrom. The audio processing module generates a representation of the audio signal. The digital audio processing module compensates the representation of the audio signal based on hearing compensation data to produce a hearing compensated audio signal. The ANR circuit receives the hearing compensated audio signal and an ANR signal. The ANR circuit further functions to adjust the hearing compensated audio signal based on the ANR signal to produce an output audio signal, wherein the ANR signal is generated based on the output audio signal.

Abstract: An apparatus comprising a transmitter configured to transmit a digital subscriber line (DSL) signal in a line from a plurality of subscriber lines according to a reference virtual noise power spectral density (PSD), wherein the reference virtual noise PSD is based on an upper bound for a worst-case far-end crosstalk (FEXT) PSD in the subscriber lines. Also included is an apparatus comprising at least one processor configured to implement a method comprising obtaining an upper bound for a worst-case FEXT PSD for a plurality of subscriber lines, obtaining a referred virtual noise PSD based on the worst-case FEXT PSD for the lines, and determining a bit-loading for the lines based on the referred virtual noise or a scaled version of the receiver-referred virtual noise.

Abstract: A combined sidetone and hybrid balance apparatus and method of operating same are disclosed. An embodiment of the present invention may provide both a hybrid balance mode of operation and a sidetone generation mode of operation within a single integrated circuit device. The functionality provided by an embodiment of the present invention may be used in both Internet protocol (IP)-based telephones and residential gateways, and may be incorporated within the IP telephone chip used in such devices. An embodiment of the present invention may be used to reduce processor demand and the cost of materials, permitting greater IP telephone or residential gateway functionality at lower cost.

Abstract: A speech enhancement device and a method for the same are included. The device includes a down-converter, a speech enhancement processor, and an up-converter. The method includes steps of down-converting audio signals to generate down-converted audio signals; performing speech enhancement on the down-converted audio signals to generate speech-enhanced audio signals; and up-converting the speech enhancement audio signals to generate up-converted audio signals.

Abstract: The present invention relates to audio signal processing, and more particularly to methods and apparatuses for emulating and controlling various features of mechanical spring reverberation in a digital audio processing system. According to certain aspects of the invention, such an emulation is performed so as to enhance or alter the characteristics of a digitally stored or processed audio signal in substantially the same manner as a mechanical spring reverberation system. In one example embodiment, the propagation of energy through a mechanical spring is simulated using dispersive waveguides, wherein left-going and right-going waves are separately processed, and the effects of dispersion and attenuation commuted to the waveguide ends. According to additional aspects, many spring reverberators contain spring elements arranged in parallel, with no coupling between springs.

Abstract: A multichannel acoustic echo reduction system is described herein. The system includes an acoustic echo canceller (AEC) component having a fixed filter for each respective combination of loudspeaker and microphone signals and having an adaptive filter for each microphone signal. For each microphone signal, the AEC component modifies the microphone signal to reduce contributions from the outputs of the loudspeakers based at least in part on the respective adaptive filter associated with the microphone signal and the set of fixed filters associated with the respective microphone signal.

Abstract: Event data messages can be provided by an interactive voice response (IVR) system to a complex events processor (CEP). The event data messages can include a Stream_ID and a series of textual elements. The Stream_ID can uniquely identify a call session between a caller and the IVR system. The series of textual elements can represent speech input provided by the caller. The CEP can create a text string from the series of textual elements of event data messages having the same Stream_ID. The text string can inherit the Stream_ID of the event data messages. The CEP can utilize user-defined business rules to process the text string. When the CEP issues an action message, the Stream_ID of the text string can be appended to the action message sent to the IVR system. The IVR system can modify the call session associated with the Stream_ID of the action message accordingly.

Abstract: An apparatus for removing an echo(es) from audio content may include a processor and memory storing executable computer code causing the apparatus to at least perform operations including receiving combined audio content including voice data associated with speech of users in a call and information including audio data provided to the users. The computer program code may further cause the apparatus to remove a first echo of a first item of voice data associated with a user(s), from the combined audio content, based in part on a prior detection of the first item of voice data. The computer program code may further cause the apparatus to remove a second echo of the audio data, from the combined audio content, based in part on a previous detection of the audio data or a previous detection of data corresponding to the audio data. Corresponding methods and computer program products are also provided.

Abstract: The present invention provides a method and apparatus for finding an estimate of the delay of a signal travelling between two points. A quantity is evaluated from the signal at a final number of time instants, at both a reference point and a reception point. The values are quantized by comparison with a threshold adapted to a typical magnitude of the quantity. If the quantized values from the reception point are shifted back by the true delay with respect to the quantized values from the reference point, then certain co-occurrences of quantized values have very low probability. Hence, the best delay estimate is that shift which yields the least number of low-probability co-occurrences.

Abstract: Processing multi-channel audio streams using one or more arrangements of single-channel components. Components that only process the near-end, or capture stream, such as noise suppression (NS) components, are limited in how they can be suitably arranged for processing multi-channel streams. However, components that process the near-end stream using one or more inputs from the far-end, or render stream, such as acoustic echo cancellation (AEC) and automatic gain control (AGC) components, are arranged in one or more of the ways suitable for use with multiple channels.

Abstract: Apparatus, methods, and machine-readable articles of manufacture enable a telephone system to distinguish between actual events and echoes. Events may include conferees' voice and DTMF commands. In selected embodiments, the system categorizes an event received on one channel as a real event if the energy of the event's waveform is greater than concurrent energy of every other channel of the same conference. Otherwise, the event is stored. When the system receives a new event of the same type on a channel of the same conference during a predetermined period, it compares the energies of the new and stored events. If the energy of the new event exceeds that of the stored event, the new event becomes the stored event, and the old stored event is ignored. At the end of the period, the stored event is categorized as a real event and forwarded to appropriate application code.

Abstract: The present invention provides a method and apparatus for finding an estimate of the delay of a signal travelling between two points. A quantity is evaluated from the signal at a final number of time instants, at both a reference point and a reception point. The values are quantized by comparison with a threshold adapted to a typical magnitude of the quantity. If the quantized values from the reception point are shifted back by the true delay with respect to the quantized values from the referenced point, then certain co-occurrences of quantized values have very low probability. Hence, the best delay estimate is that shift which yields the least number of low-probability co-occurrences.

Abstract: An apparatus comprising a digital subscriber line (DSL) transmitter configured to transmit a transmitter referred virtual noise for a tone in a subscriber line, wherein the transmitter referred virtual noise is based on a time history of a noise condition in the subscriber line. Also disclosed is an apparatus comprising at least one processor configured to implement a method comprising obtaining a transmitter referred virtual noise for a DSL tone using a plurality of noise condition measurements, and determining a DSL bitloading using the transmitter referred virtual noise.

Abstract: During playing of a streamed audio while engaging in a voice communication session, a communication device may be operable to generate a background audio during an echo cancellation operation for the voice communication session, using known information associated with the streamed audio. The streamed audio may comprise audio content that is received from one or more external sources and/or locally generated within the communication device. The known information associated with the streamed audio may comprise an echo replica corresponding to an echo associated with the streamed audio and/or a modified version of the streamed audio. In instances when a residual echo suppression is operational during the echo cancellation operation, the communication device may be operable to suppress a residual echo based on a signal level of the echo replica corresponding to the echo associated with the streamed audio and/or a signal level of the modified version of the streamed audio.

Abstract: A discrete multi-tone (DMT) transceiver communicates tones across a subscriber line. Vectoring is employed in an effort to reduce the effects of crosstalk. However, for some tones, such as tones significantly affected by radio frequency interference (RFI) or other forms of alien noise, vectoring may actually introduce distortions such that the vectoring degrades rather than improves overall signal quality. Control logic of the DMT transceiver is configured to sense when tones are affected by significant levels of alien noise and to exclude such tones from vectoring, thereby improving signal quality for such tones. The control logic also may lower the constellation densities of such tones in order to accommodate the vectoring exclusions applied to such tones.

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 CODEC circuit for POTS system comprises a digital circuit for outputting a digital control signal having a first bit width, the digital control signal being indicative of a voltage to be applied to a POTS subscriber line pair. Further, the CODEC circuit comprises a noise shaper coupled to an output of the digital circuit for generating a noise-shaped control signal and a digital-to-analog converter coupled to an output of the noise shaper, the input of the digital-to-analog converter having a second bit width being larger than 1 and smaller than the first bit width.

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: Processing multi-channel audio streams using one or more arrangements of single-channel components. Components that only process the near-end, or capture stream, such as noise suppression (NS) components, are limited in how they can be suitably arranged for processing multi-channel streams. However, components that process the near-end stream using one or more inputs from the far-end, or render stream, such as acoustic echo cancellation (AEC) and automatic gain control (AGC) components, are arranged in one or more of the ways suitable for use with multiple channels.

Abstract: A telephone system comprising an analog telephone and a full-duplex speakerphone adapter is disclosed. The analog telephone comprises an ordinary analog telephone subscriber circuit for transmitting and receiving analog signals and a handset for users. The full-duplex speakerphone adapter is coupled between the analog telephone and a central office, uses a subscriber loop interface circuit through the first telephone line to couple to the ordinary analog telephone subscriber circuit of the analog telephone and uses a telephone hybrid interface circuit through the second telephone line to couple to a wall jack to communicate with the central office.

Abstract: A voice processing apparatus includes a band dividing portion dividing a first voice signal generated by a first microphone and a second voice signal generated by a second microphone into predetermined frequency bands, a sound source segregating portion segregating an echo component of a voice emitted by a first sound source included in a voice emitted by a second sound source in each of the predetermined frequency bands based on the power of the first and second microphones, and a band synthesis portion synthesizing the first and second voice signals from which the echo component of the first sound source has been segregated by the sound source segregating portion into a voice signal including the voice emitted by the first sound source and a voice signal including the echo component of the first sound source.

Abstract: An adaptive analog echo/near-end crosstalk (NEXT) cancellation system includes a bit slicer that receives a first analog receive signal and that generates a first error control signal. A digital signal processor (DSP) receives a digital receive signal and a digital transmit signal and generates second and third error control signals. A first summing module generates a fourth error control signal by summing the second and third error control signals. A selector outputs the first error control signal during a first mode and the fourth error control signal during a second mode. An echo/NEXT cancellation module communicates with the selector and generates an estimated echo/NEXT signal based on the first error control signal and an analog transmit signal during the first mode and based on the fourth error control signal and the analog transmit signal during the second mode.