Patent classifications
G10K2210/3051
Acoustic processor having low latency
An audio processing system can include an Analog to Digital Converter structured to receive an analog input signal and convert the analog input signal to a digital input signal, a first processor coupled with the Analog to Digital Converter, the first processor including at least one programmable bi-quadratic filter chain structured to receive the digital input signal from the Analog to Digital Converter and perform audio processing on the received digital input signal at a first clock rate, and a second processor coupled with the first processor and the Analog to Digital Converter and structured to receive the digital input signal from the Analog to Digital Converter and perform audio processing on the received digital input signal at a second clock rate that is different from the first clock rate.
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.
Hybrid active noise cancellation filter adaptation
An apparatus includes a hybrid adaptive active noise control unit (HAANCU) configured to provide an anti-noise signal to an ear speaker from a reference noise signal of a reference microphone and an error signal of an error microphone, a decimator configured to decimate the reference noise signal and error signal, an adaptive hybrid ANC training unit (AHANCTU) including at least one noise cancellation filter and a filter configured to provide a feedback signal to the at least one noise cancellation, which trains parameters of the AHANCTU based on the decimated reference noise signal, the decimated error signal, and the feedback signal. The apparatus further includes a rate conversion unit configured to up-sample the parameters and update the HAANCU with the up-sampled parameters.
NOISE CANCELLATION USING ARTIFICIAL INTELLIGENCE (AI)
A method includes receiving a signal that includes noise, generating a reference signal that comprises an estimate of the noise included in the received signal, and using the reference signal to remove at least part of the noise from the received signal. The reference signal is generated by a model built using machine learning. A system includes a first apparatus that carries a signal that includes noise, and a processor based apparatus configured to execute steps including receiving the signal that includes the noise, generating a reference signal that comprises an estimate of the noise included in the received signal, and using the reference signal to remove at least part of the noise from the received signal. A storage medium storing one or more computer programs is also provided.
Apparatus and Methods for an Acoustic Noise Cancellation Audio Headset
A method and apparatus for acoustic noise cancellation (ANC) in an audio headset. A combination of an analog feedforward path and at least one digital feedforward path enables the ANC facility to operate with the shorter delay inherent in analog electronics, but with the capability of better cancellation of reflections that digital feedforward allows.
Low delay decimator and interpolator filters
Systems and methods for low latency adaptive noise cancellation include an audio sensor to sense environmental noise and generate a noise signal, an audio processing path to receive an audio signal, process the audio signal through an interpolation filter, and generate a primary audio signal having a first sample frequency, an adaptive noise cancellation processor to receive the noise signal and generate an anti-noise signal, a direct interpolator to receive the anti-noise signal and generate an anti-noise signal having the first sample frequency, and a limiter to provide clipping to reduce a number of bits in the anti-noise signal, an adder operable to combine the primary audio signal and the anti-noise signal and generate a combined output signal, and a low latency filter to process the combined output signal.
NOISE CANCELLATION WITH IMPROVED FREQUENCY RESOLUTION
A noise cancellation technique is presented with improved frequency resolution. The method includes: acquiring a digitized noise signal from an environment in which the audio signal stream is present; receiving a data sample from the digitized noise signal; appending one or more additional samples to the data sample to form a series of samples, where magnitude for each of the one or more additional samples is substantially zero; computing a frequency domain representation of the series of samples in the frequency domain; shifting the frequency domain representation of the series of samples in time using the digital processor circuit, thereby producing a shifted frequency domain representation of the series of samples; converting the shifted frequency domain representation of the series of samples to time domain to form a portion of an anti-noise signal; and outputting the anti-noise signal into the audio signal stream to abate the noise through destructive interference.
NOISE CANCELLATION WITH IMPROVED FREQUENCY RESOLUTION
A noise cancellation technique is presented with improved frequency resolution. The method includes: acquiring a digitized noise signal from an environment in which the audio signal stream is present; receiving a data sample from the digitized noise signal; appending one or more additional samples to the data sample to form a series of samples, where magnitude for each of the one or more additional samples is substantially zero; computing a frequency domain representation of the series of samples in the frequency domain; shifting the frequency domain representation of the series of samples in time using the digital processor circuit, thereby producing a shifted frequency domain representation of the series of samples; converting the shifted frequency domain representation of the series of samples to time domain to form a portion of an anti-noise signal; and outputting the anti-noise signal into the audio signal stream to abate the noise through destructive interference.
Anti-Noise Signal Generator
An anti-noise signal generator and a method of generating an anti-noise signal are presented. The anti-noise generator includes a first microphone input to receive a first sigma-delta modulated signal at a microphone sampling frequency. The first microphone input is coupled to a combiner via a first path and a second path. The combiner is adapted to combine a first filtered signal from the first path and a second filtered signal from the second path to generate the anti-noise signal. The first path includes a first digital filter adapted to operate at a filter frequency equal or greater than the microphone sampling frequency. The second path includes a second digital filter. The first digital filter may be a sigma-delta based filter that includes a sigma-delta modulator.
Digital circuit arrangements for ambient noise-reduction
A digital circuit arrangement for an ambient noise-reduction system affording a higher degree of noise reduction than has hitherto been possible. The arrangement converts the analog signals into N-bit digital signals at sample rate f.sub.0, and then subjects the converted signals to digital filtering. The value of N in some embodiments is 1 but, in any event, is no greater than 8, and f.sub.0 may be 64 times the Nyquist sampling rate but, in any event, is substantially greater than the Nyquist sampling rate. This permits digital processing to be used without incurring group delay problems that rule out the use of conventional digital processing in this context. Furthermore, adjustment of the group delay can readily be achieved, in units of a fraction of a micro-second, providing the ability to fine tune the group delay for feed forward applications.