Abstract: Embodiments include methods, systems and computer readable storage medium for a method for determining a fine direction of arrival (DOA) for a target is disclosed. The method includes receiving, by a plurality of receivers of a radar system, radar signals reflected by a target. The method further includes mitigating, by the radar system, phase shifts in the radar signals caused by a motion of the target. The method further includes determining, by the radar system, the fine DOA in response to the mitigation of phase shifts and based on the radar signals. The method further includes estimating and storing, by the radar system, a Doppler frequency based on the fine DOA.

Abstract: In some demonstrative embodiments, a simultaneous transmit and receive (STR) transceiver may include a receiver; a transmitter; a Radio Frequency (RF) echo cancellation module to generate an analog echo cancellation signal to be applied to a received RF signal via the receiver based on a transmitted RF signal via the transmitter; and a digital baseband module. The RF echo cancellation module may include a plurality of tap modulators to modulate a plurality of delayed taps of the transmitted RF signal. A tap modulator may include a plurality of phase shifters to generate a plurality of phase-shifted signals by applying a plurality of phase shifts to a delayed tap; and a plurality of multipliers to multiply the plurality of phase-shifted signals with a respective plurality of weights. The digital baseband module may determine the plurality of weights based on a plurality of subcarriers of a digital frequency-domain signal corresponding to the transmitted RF signal.

Abstract: A multi-channel audio communication system is configured to receive highly correlated input audio signals, generated as an example by multiple microphones at a far-end site. Each input audio signal is cyclically stepped through a range of discrete delay amounts, between upper and lower limits, using a step size that is a fraction of the sample period of the input audio signals. Delay cycles applied to the different input audio signals are configured to have different phases, thereby reducing the inter-signal correlation of the input audio signals. The delayed input audio signals are then played by loudspeakers. Microphone output, which may contain sound generated by the loudspeakers, is then subjected to multi-channel acoustic echo cancellation.

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 which is suitable to cope with non-linear echo paths during acoustic echo cancellation in speakerphones. Non-linear paths occur particularly in hands-free operation of, e.g., a mobile phone, due to driving the amplifier and loudspeaker in the non-linear range. The idea is to combine the commonly known one microphone approach of linear acoustic echo cancellation using an adaptive filter and a post-processor together with a multiple microphone approach using beam forming which separately removes the non-linear part of the echo.

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

Abstract: A system, 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: An oscillation-echo preventing circuit has a microphone/speaker unit and a voltage-canceling circuit for canceling voltages of audio receive signals. The microphone/speaker unit has a main body, at least two microphones, and a speaker or an earphone. The microphone seals a first inside space from an outside space. The microphone seals the first inside space from a second inside space. The speaker or the earphone seals the first inside space from the outside space. The voltage-canceling circuit cancels out the voltages of audio receive signals coming from the microphones, respectively, generating an output of minimum magnitude. Thus, the circuit can sufficiently suppress oscillation and echoing.

Abstract: Audio processing systems, such as full duplex digital telephone systems, are provided in which analog automatic level control (ALC) circuitry is employed to automatically control and stabilize the microphone output audio signal levels to a constant level for optimal digital signal processing. For example, an audio communication device (10) includes a microphone (11) to generate an audio signal, an analog automatic level control (ALC) circuit (13), coupled to an output of the microphone (11), to control a level of the audio signal output from the microphone (11) and output a level-controlled audio signal, and a DSP (digital signal processing) circuit (14) to process the level-controlled audio signal output from the ALC circuit (13). The ALC (13) allows audio signals to be captured by the microphone (11) with wide dynamic range, while stabilizing the microphone output level within a reduced dynamic range compatible with the limited dynamic range of the DSP circuitry (14) for enhanced performance.

Abstract: An uncomfortable feeling due to lack of the speech of a far end speaker and a convergence signal voice (sound) is prevented, and the echo of the far end speaker is removed. A pseudo signal generating circuit has a phase inverter (10) for inverting the phase of a signal, a pseudo hybrid circuit (11) for generating a pseudo echo having the same or substantially the same characteristic as that of the true signal, and a pseudo telephone set (5-2). By generating a pseudo echo having the phase opposite to that of the true signal and having the same or substantially the same characteristics as that of the true signal, the uncomfortable signals (echo, etc.) are cancelled out. When a telephone set is connected to a caller or when an extension transfer occurs, an adaptive filter convergence signal is generated for a predetermined constant time only during “no-voice (sound) (silence)” of the far end speaker's signal.

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: Disclosed is a non-linear echo canceller and method for cancelling echo during full duplex communication in a hands free communication system. An input signal from a far-end talker and an input signal from the output from an echo canceller are received. K spectral subbands are created for each input signal. The spectral echo residual power at each subband is estimated and compared to a clean signal power to calculate a signal to echo ratio. Gains are calculated based on each calculated ratio and non-linear echo is cancelled based on the calculated gains.

Abstract: A field-mounted process device with multiple isolated connections includes a connection that can be an input or an output. The given input or output can couple to multiple sensors or actuators, respectively. The process device can be wholly powered through its communication I/O port. The process device includes a controller adapted to measure one or more characteristics of sensors coupled to an input connection and to control actuators coupled to an output connection. The controller can be further adapted to execute a user generated control algorithm relating process input information with process output commands.

Abstract: The present invention provides mechanisms for removing duplicate signals during a communication session. More particularly an over-air signal is received at a given endpoint and a stream of electronic data is monitored at the same endpoint for a duplicate to the over-air signal. If a duplicate is detected in the electronic data stream, it is removed so that only one occurrence of the signal is received at the endpoint.

Abstract: Techniques for suppressing large echo due to nonlinearity, instable adaptive filter, and so on. An echo cancellation system includes an echo canceller unit, a nonlinear echo detection unit, and a nonlinear echo suppression unit. The echo canceller unit (e.g., an adaptive filter) receives a reference signal and a near-end signal, derives an echo estimate signal based on the reference and near-end signals, cancels a portion of echo in the near-end signal based on the echo estimate signal, and provides an intermediate signal having residual echo. The nonlinear echo detection unit detects for large nonlinear echo in the echo estimate signal. The nonlinear echo suppression unit suppresses large nonlinear echo in the first intermediate signal based on a variable gain, which may be adjusted lower if large nonlinear echo is detected and higher otherwise.

Abstract: A method and apparatus for adapting dual filters is disclosed. In one aspect, a method may include transforming a signal, adapting a first adaptive filter based on the transformed signal, estimating a delay of an impulse response based on the adaptation of the first filter, delaying a signal based on the estimated delay, and adapting a second adaptive filter based on the delayed signal. In one aspect, an echo or other unwanted signal may be reduced or cancelled based on the adaptation of the second filter.

Abstract: A VoIP telephone operates in an IP environment wherein at least a portion of the signaling from a remote gateway is included within a media session real time protocol stream. The VoIP telephone comprises a network communication system for encapsulating data into IP frames for exchange with remote devices over a frame switched network. A system client application is coupled to the network communication system and exchanges call set up messages with a remote VoIP gateway to establish a media channel for the exchange of the real time protocol streams. The system client also provides VoIP status signals to a presentation module. A dialog system is coupled to the network communication system and: i) translates frames of compressed digital audio data originated from a remote device to recreate remote voice band; ii) detects and compresses local voice band for transmission to the VoIP gateway; iii) detects in band signaling within the voice band; and iv) generates in band status signals to the presentation module.

Abstract: A circuit arrangement for the analogue suppression of echoes, as in particular can be used in a hybrid-circuit for DSL-transmission systems, comprises a replica (8) for emulating the behaviour of the transmission line (17). In addition, a circuit (3, 4) for emulating the behaviour of the transmitter (13) is provided, which comprises at least one lowpass (3, 4). Furthermore, a replica (9, 10) for emulating the behaviour of bridge taps (14) can also be provided, which comprises at least one bandpass (9, 10). Additionally, a replica (19) for emulating the behaviour of the line driver (1) can also be provided.

Abstract: In a microphone signal, the signal component corresponding to, e.g., echo is suppressed using an echo control scheme that estimates the spectral envelope of the echo signal, without having to estimate the waveform for the echo signal. In one embodiment, the input signal (to be applied to a loudspeaker) and the microphone signal are spectrally decomposed into multiple subbands, where echo suppression processing is independently performed on each subband. The echo control of the present invention can be implemented with substantially reduced (1) computational complexity and (2) phase sensitivity, as compared to traditional acoustic echo cancellation, in which the waveform for the echo signal is estimated.

Abstract: A method and apparatus for real-time derivation of precise digital clock edges and synchronous logic samples from a digital signal having a clock channel and at least one data channel acquires a plurality of temporally offset analog samples during each of a sequence of sample periods and from consecutive samples where there is a logic level transition estimates an edge time. From the edge times for the clock channel an offset is added and applied to the at least one data channel to determine the synchronous logic samples for the data channel at each offset clock edge time.

Abstract: A method and apparatus enables echo reduction in a full duplex transceiver system. A replica current is subtracted from a receiver via a first differential circuit path that adaptively matches a time constant associated with a second differential circuit path that connects the receiver with an external data line.

Abstract: A speaker-phone that is operable to perform system change detection, double-talk detection and if desired, optimal step-size control. The invention differentiates between system change and double-talk and is operable to these two situations differently. The invention does not rely solely on the energy measurements that are employed by conventional speaker-phones. Highly reliable system change detection is offered when compared to conventional speaker-phones. The direction and the amplitude of the system change is measured with very good detection probability in the invention. For double-talk detection, the invention does not rely on assumptions that the far-end echoes are of lower amplitude that the near-end signal in performing double-talk detection. The invention performs double-talk detection within a large diversity of echo-paths and background noise levels. A designer may tune the speaker-phone system to ensure proper double-talk detection near low levels of the near-end signal.

Abstract: An echo sound signal suppressing apparatus for suppressing an echo sound signal to ensure speech communication between a far-end speaker and a near-end speaker, comprising: first power value calculating means for calculating the power value of the second sound signal; second power value calculating means for calculating the power value of the echo replica sound signal produced by the echo replica sound signal producing means; echo sound suppressing means for suppressing the echo sound difference signal component outputted from the difference signal producing means to a minimum level with the power value of the second sound signal calculated by the first power value calculating means and the power value of the echo replica sound signal calculated by second power value calculating means with the near-end speaker sound signal component being allowed to pass therethrough, the echo sound suppressing means having an output terminal to the communication line to output the near-end speaker sound signal component unde

Abstract: In handsfree communications devices including a linear echo or noise canceler, both analog and digital gain factors are adjusted in response to user volume control inputs, the analog gain factor being frozen during active communications (e.g., during telephone calls). By providing dual user controls, one of which is disabled when a call is in progress, embodiments of the invention provide the wide dynamic range required in most handsfree applications without degrading performance of the linear echo canceler.

Abstract: There is disclosed an audio terminal (10) and a method for operating in uncontrolled audio environments, the audio terminal (10) having an echo suppression unit (20) for reducing acoustic feedback (18). The echo suppression unit (20) includes a learner (22) for learning an audio environment of the audio terminal (10) and a control unit (21) for controlling the acoustic feedback (18) in accordance with the audio environment of the audio terminal (10).