Abstract: A variable update step size is determined in proportion to a magnitude ratio or magnitude difference between a first residual signal and a second residual signal. The first residual signal is obtained by using adaptive filter coefficient sequence, where the adaptive filter coefficient sequence has been obtained in previous operations of the adaptive equalizer. The second residual signal is obtained by using a prior update adaptive filter coefficient sequence, where the prior update adaptive filter coefficient sequence is obtained by performing a coefficient update with an arbitrary prior update step size on the adaptive filter coefficient sequence having been obtained in previous operations of the adaptive equalizer.

Abstract: Embodiments of the invention may be used to implement a rate converter that includes: 6 channels in forward (audio) path, each channel having a 24-bit signal path per channel, an End-to-end SNR of 110 dB, all within the 20 Hz to 20 KHz bandwidth. Embodiment may also be used to implement a rate converter having: 2 channels in a reverse path, such as for voice signals, 16-bit signal path per channel, an End-to-end SNR of 93 dB, all within 20 Hz to 20 KHz bandwidth. The rate converter may include sample rates such as 8, 11.025, 12, 16, 22.05, 24, 32 44.1, 48, and 96 KHz. Further, rate converters according to embodiments may include a gated clock in low-power mode to conserve power.

Abstract: In digital video image encoding and decoding, a filter type is selected based on symmetrical properties of the images and coefficient values of an interpolation filter are calculated based on the selected filter type. Coefficient values, filter tap-length and selected filter-type are provided in the encoded video data. Coefficient values are also calculated based on a prediction signal representative of the difference between a video frame and a reference image. The prediction signal is calculated from the reference image based on a predefined base filter and motion estimation performed on the video frame. The predefined base filter has fixed coefficient values. Coefficient values are selected from interpolation of pixel values in a selected image segment in the video frame. Symmetry properties of images can be a vertical symmetry, a horizontal symmetry and a combination thereof, so that only a portion of the filter coefficients are coded.

Abstract: A low noise receiver includes a downconverter configured to receive a radio frequency (RF) signal, the downconverter comprising a switching architecture configured to generate a plurality of output phases based on a respective plurality of local oscillator (LO) signals, a differencing circuit configured to combine the plurality of output phases such that an nth output phase is differenced with an (n+K)th output phase, resulting in gain-added output phases, and a summation filter configured to receive the gain-added output phases and configured to combine the gain-added output phases such that a response of the receiver effectively reduces odd harmonics of the RF signal.

Abstract: An echo cancellation system performs audio beamforming to separate audio input into multiple directions (e.g., target signals) and generates multiple audio outputs using two acoustic echo cancellation (AEC) circuits. A first AEC removes a playback reference signal (generated from a signal sent a loudspeaker) to isolate speech included in the target signals. A second AEC removes an adaptive reference signal (generated from microphone inputs corresponding to audio received from the loudspeaker) to isolate speech included in the target signals. A beam selector receives the multiple audio outputs and selects the first AEC or the second AEC based on a linearity of the system. When linear (e.g., no distortion or variable delay between microphone input and playback signal), the beam selector selects an output from the first AEC based on signal to noise (SNR) ratios. When nonlinear, the beam selector selects an output from the second AEC.

Abstract: An echo cancellation system that uses a combined reference signal using a playback reference signal and an adaptive reference signal. The playback reference signal is generated from a playback signal sent to a loudspeaker and the adaptive reference signal is generated using beamforming on microphone inputs corresponding to audio received from the loudspeaker. The system applies a low pass filter to the playback reference signal and applies a high pass filter to the adaptive reference signal to generate the combined reference signal. The system may remove the combined reference signal from target signals associated with the microphone inputs to isolate speech included in the target signals.

Abstract: A node for a communication system that is configured to communicate with a mobile node via a first connection and to communicate with a third node via a first forwarding path is disclosed. After a procedure for handover of the first connection to a second connection between the mobile node and a target network node is initiated, data is communicated from the node to the third node for transmission to the target network node via a second forwarding path. The third node is configured to receive the data and to forward it to the target network node.

Abstract: Embodiments of the present invention provide a virtual multicarrier design for orthogonal frequency division multiple access communications. Other embodiments may be described and claimed.

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: An impedance adjustment method for a communication device, wherein the communication device has a plurality of impedance paths for selection, includes: selecting an initial impedance path; and utilizing a predetermined algorithm to examine a portion of the plurality of impedance paths by starting from the initial impedance path for selecting an optimized impedance path for the communication device. A delay capacitance adjustment method for a communication device, wherein the communication device has a plurality of delay capacitance paths for selection, includes: selecting an initial delay capacitance path; and utilizing a predetermined algorithm to examine a portion of the plurality of delay capacitance paths by starting from the initial delay capacitance path for selecting an optimized delay capacitance path of the communication device.

Abstract: A pre-distortion coefficient corresponding to a transmit power currently set by a transmitting end is adjusted. A signal received quality report which is obtained according to the pre-distortion coefficient before adjustment is analyzed and fed back by a receiving end and a signal received quality report that is obtained according to the pre-distortion coefficient after adjustment and fed back by the receiving end. The analysis determines a variation situation of signal received quality of the receiving end. If the signal received quality of the receiving end becomes better, the pre-distortion coefficient after adjustment is used as the pre-distortion coefficient corresponding to the transmit power currently set by the transmitting end. The process then returns to the step of adjusting a pre-distortion coefficient corresponding to a transmit power currently set by a transmitting end.

Abstract: A system for detecting motion comprising a first speaker, a first microphone separated from the first speaker by a distance D1, a sound generator, an echo parameter measurement device and an echo parameter monitor, wherein the echo parameter monitor stores two or more sequential echo parameters and generates a motion indicator if the two or more sequential echo parameters indicate a change in an acoustic echo path that exceeds a predetermined threshold.

Abstract: A processing system can include a tracking microphone array; audio tracker circuitry connected to the tracking microphone array to track an audio source based on an audio input from the array; communication microphones; and a processor. The processor can include audio circuitry to receive an audio input from the communication microphones and process the audio input to apply one or more of acoustic echo cancellation (AEC) and acoustic echo suppression (AES) processing to the audio input. The processor can further include calculating circuitry to calculate a ratio of signal power after and before the AEC and/or the AES processing, and control circuitry to generate an acoustic echo presence indication based on the ratio calculated by the calculating circuitry. The processor can transmit, via transmitting circuitry, the acoustic echo presence indication to an audio tracking device via a data communication channel between the processor and the audio tracker.

Abstract: A method for determining a sampling time of a signal, the method including: determining a candidate sampling time of a signal for input to a sampling circuit; determining a resultant sampling time at which the sampling circuit samples the signal when input with the candidate sampling time; and determining a sampling time of the signal based on a noise shaping of a difference between the resultant sampling time and the candidate sampling time.

Abstract: The present invention relates to a method and an apparatus for efficiently controlling an uplink control signal of user equipment in an environment where a macro cell and a repeater exist together, and the method for controlling uplink transmission power of the user equipment in the wireless communication system, according to the present invention, comprises: an obtaining step for receiving a reference signal from a base station and obtaining a channel state information value; a determining step for determining whether a scaling condition is satisfied by using the channel state information value that is obtained; a calculating step for calculating a transmission power on the basis of scheduling information when the scaling condition is satisfied; and a transmitting step for transmitting data by using the transmission power that is calculated.

Abstract: In some aspects, techniques for wireless signal decoding are described. In some instances, a number of superposition codes are computed based on a set of orthogonal binary codes. The set of orthogonal binary codes includes multiple subsets; each superposition code corresponds to one of the subsets and includes a combination of the binary codes in the corresponding subset. One of the subsets is selected based on a correlation between a wireless signal and the superposition code that corresponds to the subset. One of the binary codes is then selected from the selected subset based on a correlation between the wireless signal and the binary code. The wireless signal is decoded based on the selected binary code.

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

Abstract: Methods, apparatuses, systems, and software are disclosed for providing a phone or other audio communication device with a fixed-path acoustic echo cancellation feature that compensates for a fixed-path acoustic coupling caused by transmission of sound from one or more speakers to one or more microphones through, for instance, the phone body itself.

Abstract: Apparatus and methods disclosed herein implement an RF receive-band filter at a receive chain input of a wireless base station with a co-located transmitter and receiver. The RF receive-band filter includes an adaptive filter component to perform filtering operations on samples of a digital baseband or intermediate frequency signal x(n) from a transmit chain associated with the wireless base station. An adaptive filter transfer function is determined in real time such that samples of the baseband transmit signal x(n) are transformed into a cancellation baseband signal z(n). The digital cancelation baseband signal z(n) is then digital-to-analog converted and the resulting analog baseband signal z(t) is up-converted to obtain a subtractive RF cancelation signal c(t). C(t) is summed with a desirable received signal RF component r(t) and an undesirable transmitter leakage RF signal component l(t) appearing at the input to the base station receiver.

Abstract: A filter enhancer provides greater performance in a passive filter. The passive filter has an input terminal for receiving a signal in a first frequency band and an output terminal for providing a signal in a second frequency band. The filter enhancer includes: (a) a first canceller circuit coupled between input terminal and output terminal, the first canceller circuit including a first finite impulse response (FIR) filter for attenuating signals in the first frequency band; and (b) a control circuit coupled to output terminal and first canceller circuit for providing adaptive coefficients for configuring the first FIR filter. The first FIR filter may be implemented by either a digital FIR filter or an analog FIR filter. The filter enhancer may further include a second canceller circuit coupled between input terminal and output terminal, the second canceller circuit including a second FIR filter for attenuating signals in the second frequency band.

Abstract: A method and system for validating SIPSI of a broadcast table in an electronic device is provided. The method includes receiving a first cache signature and retrieving a second cache signature from a memory. The method also includes identifying differences between the first cache signature and the second cache signature. Further, the method includes performing one of updating the broadcast table and updating a portion of the broadcast table. The system includes an electronic device for validating SIPSI of a broadcast table. The electronic device includes a broadcast receiver for receiving a first cache signature and a memory for storing a second cache signature. The electronic device also includes a processor for identifying differences between the first cache signature and the second cache signature. Further, the electronic device includes the processor for performing one of updating the broadcast table and updating a portion of the broadcast table.

Abstract: Using switching technologies to duplicate packets of a digital stream (e.g., digital video stream) sent from one workstation to multiple recipient workstations, where the switching technologies enable the multiple streams sent from the switch to the recipient workstations to be generated from a single digital stream sent from the sending workstation to the switch. Data units, such as video data units, may be transmitted by using a switch to enable receipt of a stream of data units including a payload portion and an attribute portion from at least two conferencing participants. The switch is used to duplicate at least a subportion of the payload portion of a data unit within the stream of data units, and to enable access to the duplicated subportion of the data unit by two or more conferencing participants.

Abstract: An apparatus and a method for estimating a channel in a broadband wireless access system. A receiving apparatus includes a first channel estimation filter having a tap coefficient which is channel-adaptively variable, the first channel estimation filter acquiring a channel impulse response by filtering received pilot symbols, a second channel estimation filter having a fixed tap coefficient, the second channel estimation filter acquiring a channel impulse response by filtering received pilot symbols, and a selector which selects the operation of one of the first and the second channel estimation filters according to channel correlativity.

Abstract: Conserving power for coded transmissions comprises ceasing to process parity packets once associated data packets are deemed correct or corrected. Once data packets are deemed correct or corrected, the receiving unit can shut off during the transmission of parity packets.

Abstract: A communication device has a controller operatively connected to at least a first transceiver and a second transceiver, wherein the first transceiver receives signals on one or more channels within a first frequency band and the second transceiver transmits signals on one or more channels within a second frequency band, wherein the first and second frequency bands are adjacent one another so that each of the first and second frequency bands has an adjacent border and a nonadjacent border. Coexistence between the first and second transceivers is achieved by adjusting receive and/or transmit filters associated with the transceivers to create a guard band that is located more in the first frequency band if the second transceiver is using frequencies close to its adjacent border, and a guard band that is more in the second frequency band if the second transceiver is not using frequencies close to its adjacent border.

Abstract: A communication control apparatus including: a count portion that counts the number of received data; and a discard portion that is provided at an upstream side of a control portion on a forwarding route of the received data, and discards received data according to a counted value by the count portion.

Abstract: The present invention provides a technique for automatically establishing a virtual circuit from one edge device to another through a core network. The virtual circuit may comprise one or more pseudowire segments. These pseudowire segments extend between adjacent edge and core devices within the core network. Pseudowire segments through the core network are automatically stitched together as a setup message is forwarded from a first edge device to a second edge device through the core network. A virtual circuit is established in one direction, and packets are forwarded in an opposite direction. With the present invention, a virtual circuit comprising any number of pseudowire segments, as well as two attachment circuits, may be established in an automatic and efficient manner upon being initiated by the first edge device.

Abstract: Disclosed are various embodiments providing carrier separation to effectuate a dual carrier operation of a wireless communication device. Processing circuitry may acquire a single carrier signal based on a determined time slot boundary, the single carrier signal comprising primary carrier data. The processing circuitry may receive a dual carrier radio frequency signal based on the determined time slot boundary, the dual carrier radio frequency signal comprising the primary carrier data and secondary carrier data. The processing circuitry may then generate a baseband dual carrier signal by demodulating the dual carrier radio frequency signal. The processing circuitry may separate the primary carrier data from the baseband dual carrier signal by filtering the baseband dual carrier signal. The processing circuitry may also separate the secondary carrier data from the baseband dual carrier signal by filtering the baseband dual carrier signal.

Abstract: A radio transceiver including a transmitter and a receiver. A filter coupled to an output of the transmitter, the filter has one or more inductors and one or more capacitors, where the filter is tuned by varying one or more capacitance values of the one or more capacitors in the filter to tune the filter. The one or more capacitors are Barium Strontium Titanate (BST) ceramic integrated capacitors that are each tuned by application of a bias voltage thereto. A filter tuner circuit is configured to apply the bias voltage to the one or more capacitors, where the bias voltage is initially established as a stored initial value, and where the bias voltage is refined by a proportional-integral-derivative (PID) controller configured to optimize a power within the transceiver. This abstract is not to be considered limiting since various implementations may incorporate more, fewer or different elements.

Abstract: A method including receiving in a first network interface of a router a first request for a first type of multicast traffic in a first multicast routing protocol, storing in the router a second type of multicast traffic, transmitting from the router using a second multicast routing protocol requests for the first and second types of multicast traffic, receiving in the router the first and second types of multicast traffic comprising first and second IP packets, respectively, modifying in the router the IP source address and/or the IP multicast destination address of the second IP packets so that the addresses of the second IP packets are the same as the first IP packets of the first type of multicast traffic and transmitting the first IP packets and the modified second IP packets via the first network interface, where the router transmits in a first time interval the first IP packets, transmits in a second time interval the modified second IP packets, and transmits in a third time interval the remaining porti

Abstract: A transceiver system is disclosed. The transceiver system comprises a first transceiver physical layer circuit (PHY) having a first plurality of channels and a second transceiver PHY disposed adjacent the first transceiver PHY and having a second plurality of channels. Filter circuitry is coupled between at least one of the plurality of first channels and at least one of the plurality of second channels.

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

Abstract: In one embodiment, a signal processing method is disclosed. The method can perform filter processing of convoluting a tap coefficient in a first signal sequence to generate a second signal sequence. The method can subtract the second signal sequence from a third signal sequence to generate a fourth signal sequence. The third signal sequence includes an echo signal of the first signal sequence. The method can correct the tap coefficient in accordance with an amount of correction determined using a function. The function includes at least one of a first region and a second region, and has values limited. The first region is included in a negative value region of the fourth signal sequence. The second region is included in a positive value region of the fourth signal sequence.

Abstract: A frequency-domain based echo and NEXT canceller is provided. The canceller uses log2 encoding to precondition the error signal representing the echo. An improved gradient constraint is applied on at least a portion of a full weight vector in a least-mean-square algorithm. The least-mean-square algorithm is used to compute filter coefficients. The filter coefficients are multiplied by a frequency-domain data vector using a frequency-domain multiplier to generate frequency-domain output vector.

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

Abstract: A packetized streaming media delivery network carries many “streams” of differing media content. They often are from multiple sources and of different media types. The invention consists of a scalable hardware and/or software computing element resolving the network traffic into its individual streams for focused, simultaneous, and continuous real-time monitoring and analysis. The monitoring and analysis consists of delay factor and media loss rate which measure the cumulative jitter of the streaming media within the delivery network and the condition of the media payload. These measurements form a powerful picture of network problem awareness and resolution. The delay factor objectively indicates the contribution of the network devices in the streams' path, allowing for both problem prediction and indication. In one example, tapping a packetized network at various locations allows for correlation of the same-stream performance at various network points to pinpoint the source(s) of the impairment(s).

Abstract: Aspects of a method and system for Ethernet Switching, Conversion, and PHY optimization based on link length in Audio/Video Systems are provided. In this regard, the length of a link over which a first communication device communicates with a second communication device may be determined and Ethernet PHY's residing in the communication devices may be configured based on the determination. One or both of the communication devices may comprise an Ethernet switch operable to switch packetized audio and/or video content. The packetized audio and/or video may be switched between the configured Ethernet PHY and one or more other Ethernet PHYs. Audio and/or video data may be received via the one or more corresponding Ethernet PHYs, the received audio and/or video data may be reformatted to be compatible with an audio and/or video interface, and the reformatted audio and/or video data may be transmitted via the audio and/or video interface.

Abstract: To assist with the detection of unfiltered device(s), a system observes how the received noise changes between two links as the transmit signal is changed. Harmful unfiltered nonlinear devices will generate significant noise that depends on the transmitted signals; therefore, this additional noise can be quantified to some extent by comparing the observed noise for two different transmit signals. The total noise can be determined from the SNR if the received signal is known. The received signal may be read directly in some non-standard systems, or it may be determined from the known transmit signal and channel attenuation, which is sometimes the case in standard-compliant links, but often with a relatively large error. To circumvent this problem, certain embodiments of this invention only consider the change in noise between two links with the same channel attenuation. This differential comparison makes it unnecessary to accurately know the channel attenuation.

Abstract: A digital equalizer with a reduced number of multipliers for correction of the frequency responses of an interleaved ADC is disclosed. An exemplary interleaved analog to digital converter with digital equalization includes a composite ADC including M time interleaved sub-ADC, a demultiplexer, samples repositioning unit, a first PreFIRs transformer, a second PreFIRs transformer, K double buffer FIR filters, a PostFIRs transformer, a samples sequence restoration unit, and a multiplexer, coupled in series and providing an equalized, frequency response-corrected output.

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

Abstract: 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: Methods and systems for dual digital microphone processing in an audio CODEC are disclosed and may include demultiplexing one or more received time-multiplexed digital audio signals from one or more digital microphones, and separately processing each of the demultiplexed digital audio signals. The digital microphones may include microelectromechanical (MEMS) microphones. The demultiplexed digital audio signals may be level-converted, downshifted, and/or filtered. The filtering may include a finite impulse response (FIR) filter. A sampling rate of the one or more demultiplexed digital audio signals may be converted by repeating the demultiplexed digital audio signals. Audio beamforming and/or diversity processing may be performed utilizing the digital microphones.

Abstract: A divergence detection device for an adaptive system comprising means for calculating the energy of an input signal of the system, means for calculating the energy of an output signal of the system and means for analysing the energy of the input signal and the energy of the output signal so as to detect a rise in the energy of the output signal, relative to the energy of the input signal, in case of divergence of the system.

Abstract: A system, method, and program product are provided for filtering sound from a selected application on a computer without interrupting voice communications on the computer. The method comprises: monitoring a selected program for an outgoing digital audio signal from a selected application; detecting said digital audio signal; and filtering an analog microphone input with the digital audio signal.

Abstract: Methods and corresponding systems in an adaptive filter include calculating a signal estimator output using filter coefficients, and calculating an error signal. Next, a coefficient threshold is determined. Thereafter, for each filter coefficient, a first step size is assigned to filter coefficients with a magnitude less than the coefficient threshold, and a second step size is assigned to filter coefficients with a magnitude greater than or equal to the coefficient threshold. Finally, the filter coefficients are updated using the first and second step sizes and the error signal. The coefficient threshold can be selected as the average of the magnitudes of the filter coefficients. Alternatively, the coefficient threshold can be selected as the Mth largest of the filter coefficients ranked in order of magnitude. In one embodiment, the first step size can be less than one and the second step size can be greater than one.

Abstract: Methods and apparatus for canceling distortion in full-duplex transceivers are disclosed. Some example methods to reduce distortion in a full-duplex transceiver include generating a first digital signal, generating a first analog signal based on the first digital signal for transmission over a full-duplex channel, receiving a second analog signal via the full-duplex channel, and generating a second digital signal based on the second analog signal, wherein the second digital signal includes coupling distortion based on the first analog signal. The example methods further include generating an adaptive filter signal based on the first digital signal, and reducing the coupling distortion from the second digital signal by subtracting the adaptive filter signal from the second digital signal.

Abstract: Provides is an embedded code-excited linear prediction speech coding/decoding apparatus and method that can deal with the capacity change of speech transmission channel by modeling an error signal not coded at a core speech coder based on a transmission rate in a multiple pulse search mode or gain compensation mode and then transmitting it in an optimum mode. The apparatus includes a core speech coding unit for coding an input speech signal with spectral envelop and an excitation signal, a transmission rate determination unit for allocating the number of bits additionally allowed depending on a capacity of a transmission channel, and an embedded excitation signal coding unit for coding a residual excitation signal that is not coded in the core speech coding unit based on the number of additionally allowed bits using one of a multiple pulse excitation coding mode and a gain compensation mode.

Abstract: Embodiments of a method and apparatus of reducing transmit signal components of a receive signal of a transceiver are disclosed. One embodiment of an apparatus includes a transceiver that simultaneously transmits a transmit signal and receives a receive signal. The transceiver includes a transmit DAC that generates the transmit signal based on a transmit digital signal stream. The transmit DAC includes a plurality of transmit DAC circuit elements, and a plurality of transmit DAC switches that control which of the plurality of transmit DAC circuit elements contribute to generating the transmit signal. The transceiver additionally includes an echo cancellation DAC that generates an echo cancellation signal based on the transmit digital signal stream.

Abstract: A second output transmission signal (“TX2”) added to a line driver is a scaled version of the main output transmission signal (“TX1”). TX2 is scaled from TX1 by a variable scale factor K. An adaptive hybrid circuit subtracts TX1 and TX2 from a line signal carrying both a line transmission signal and a line received signal (“RX”). A programmable impedance Ztune is coupled between the TX2 output of the line driver and the RX output of the adaptive hybrid circuit. A transmission echo in the output RX signal is measured. K and Ztune are then adaptively tuned to minimize the transmission echo. The hybrid in this case becomes a 4-port network, one port specifically added to adaptively cancel the transmission echo in the RX output of the adaptive hybrid circuit. Alternatively, the hybrid may be a 3-port hybrid including variable impedances to cancel the line transmission signal.

Abstract: A mobile communication device capable of communicating with a remote communication device includes a communication unit, a microphone, an input unit, a control unit, and a switching unit. The communication unit is used for generating a remote audio signal according to remote audio information received from the remote communication device. The microphone is used for generating a local audio signal according to a local sound. The input unit is used for receiving a trigger signal. The control unit is configured for generating a switching signal in response to the trigger signal. The switching unit is configured for transmitting the remote and local audio signals to the remote communication device via the communication unit alternately according to the switching signal.