Patent classifications
H03H21/00
SUBBAND ADAPTIVE FILTER FOR SYSTEMS WITH PARTIALLY ACAUSAL TRANSFER FUNCTIONS
A noise reduction system includes sensors configured to generate an input signal, an adaptive filter configured to represent a transfer function of a path traversed by the input signal, one or more processing devices, and one or more transducers. The processing devices receive the input signal and generate an updated set of filter coefficients of the adaptive filter by separating the input signal into frequency subbands; determining for each subband, coefficients of a corresponding subband adaptive module; and combining the coefficients of multiple subband adaptive modules. Determining the coefficients of the corresponding subband adaptive module includes selecting a subset of a precomputed set of filter coefficients of the adaptive filter. The processing devices process a portion of the input signal using the updated set of filter coefficients of the adaptive filter to generate an output that destructively interferes with another signal traversing the path represented by the transfer function.
Compensating for channel distortion during contactless communication
Systems, methods, and devices are provided for compensating for distortion of a contactless communication channel. The electronic device may include a radio frequency system that itself includes antenna to transmit and receive data using near-field communication (NFC) and an NFC signal processing circuitry. The NFC signal processing circuitry may receive an NFC signal via a communication channel formed between the electronic device and another electronic device and may determine a baseband reference waveform associated with the electromagnetic NFC signal and may determine an error between a portion of the electromagnetic NFC signal and the baseband reference waveform. Furthermore, the NFC signal processing circuitry may determine whether the error is outside of an acceptable error threshold range and, in response to the error being outside of the acceptable error threshold range, train a filter response of the NFC signal processing circuitry to estimate the communication channel.
Adaptive harmonic cancellation
Disclosed systems and methods relate to an adaptive harmonic cancellation circuit for communication. The adaptive harmonic cancellation circuit includes a harmonic generator circuit configured to generate a reference harmonic of an interference signal. The adaptive harmonic cancellation circuit includes a harmonic prediction circuit coupled to the harmonic generator circuit. The harmonic prediction circuit is configured to receive an input signal including a target signal at a frequency and a radiated harmonic of the interference signal. The harmonic prediction circuit is configured to generate a predicted harmonic of the interference signal by modifying the reference harmonic of the interference signal to match the radiated harmonic of the interference signal in the input signal. The adaptive harmonic cancellation circuit includes a cancellation circuit coupled to the harmonic prediction circuit. The cancellation circuit is configured to obtain the target signal at the frequency by subtracting the predicted harmonic from the input signal.
Transmitter, receiver and a method for digital multiple sub-band processing
Highly efficient digital domain sub-band based receivers and transmitters.
SYSTEMS, APPARATUSES AND METHODS FOR ADAPTIVE NOISE REDUCTION
An apparatus includes a sensor module configured for receiving sensed information indicative of a sensed signal. The sensed signal includes a source signal component and a source noise component. The apparatus also includes a reference module configured for reference information indicative of a reference signal. The reference signal also includes a reference noise component. The apparatus also includes a filter module configured as a fixed lag Kalman smoother. The filter module is configured for adaptively filtering the reference signal to generate an estimate of the source noise component. The apparatus also includes a processing module configured for calculating an output signal based on the sensed signal and the estimate of the source noise component. The apparatus also includes an interface module configured for transmitting an indication of the output signal. The filter module is further configured for, based on the output signal, tuning the Kalman smoother.
Precision digital to analog conversion in the presence of variable and uncertain fractional bit contributions
This disclosure describes systems, methods, and apparatus for a digital-to-analog (DAC) converter, that can be part of a variable capacitor and/or a match network. The DAC can include a digital input, an analog output, N contributors (e.g., switched capacitors), and an interconnect topology connecting the N contributors, generating a sum of their contributions (e.g., sum of capacitances), and providing the sum to the analog output. The N contributors can form a sub-binary sequence when their contributions to the sum are ordered by average contribution. Also, the gap size between a maximum contribution of one contributor, and a minimum contribution of a subsequent contributor, is less than D, where D is less than or equal to two time a maximum contribution of the first or smallest of the N contributors.
Short link efficient interconnect circuitry
Systems and methods for electronic devices including two or more semiconductor devices coupled via an interconnect. The interconnect includes multiple lanes each having a link between the first and second semiconductor devices. One or more lanes of the multiple lanes each include clock and data recovery circuitry to perform full clock and data recovery. One or more other lanes of the multiple lanes each do not include clock and data recovery circuitry and instead includes a phase adjustment and clock multiplier circuit that is slave to clock and data recovery circuitry of the one or more lanes.
Subband adaptive filter for systems with partially acausal transfer functions
A noise reduction system includes sensors configured to generate an input signal, an adaptive filter configured to represent a transfer function of a path traversed by the input signal, one or more processing devices, and one or more transducers. The processing devices receive the input signal and generate an updated set of filter coefficients of the adaptive filter by separating the input signal into frequency subbands; determining for each subband, coefficients of a corresponding subband adaptive module; and combining the coefficients of multiple subband adaptive modules. Determining the coefficients of the corresponding subband adaptive module includes selecting a subset of a precomputed set of filter coefficients of the adaptive filter. The processing devices process a portion of the input signal using the updated set of filter coefficients of the adaptive filter to generate an output that destructively interferes with another signal traversing the path represented by the transfer function.
Method and apparatus for adaptive signal processing
A method for adaptive signal processing is provided. In the method, a second vector is obtained by initializing a first vector without regularization of a cost function. The cost function is regularized with the first vector and the second vector as variables. The first vector is updated based on an input signal, according to the regularized cost function. Then, an output signal is provided based on the updated first vector. The second vector is updated based on the update of the first vector. An apparatus for adaptive signal processing is provided accordingly. The method and the apparatus are well compatible with existing adaptive signal processing. The convergence coefficients of the adaptive filter system become more stable. Moreover, impact of an extra penalty added to the cost function on a bias can be minimized, and the increased complexity of the system is very limited.
Advanced audio feedback reduction utilizing adaptive filters and nonlinear processing
Traditional audio feedback elimination systems may attempt to reduce the effect of the audio feedback by simply scaling down the audio volume of the signal frequencies that are prone to howling. Other traditional feedback elimination systems may also employ adaptive notch filtering to detect and notch the so-called singing or howling frequencies as they occur in real-time. Such devices may typically have several knobs and buttons needing tuning, for example: the number of adaptive parametric equalizers (PEQs) versus fixed PEQs; attack and decay timers; and/or PEQ bandwidth. Rather than removing the singing frequencies with PEQs, the devices described herein attempt to holistically model the feedback audio and then remove the entire feedback signal. Two advantages of the devices described herein are: 1.) the system can operate at a much larger loop-gain (and hence with a much higher loudspeaker volume); and 2) setup is greatly simplified (i.e., no tuning knobs or buttons).