Systems and methods herein provide for adaptive subband compression of power signals in a power system. In one embodiment, a system includes an encoder is operable to partition sensor measurements into frequency subbands (e.g., including an interharmonic subband), centered at integer multiples of the power system's fundamental frequency (e.g., 50 Hz or 60 Hz). The encoder may also be operable to detect active subbands, and to compress the at least one active subband. The system also includes a data concentrator operable to transmit the at least one compressed subband to a processor for analysis. The system also includes a decoder at a processing location (a substation, a concentrator, or the control center) operable to parse the compressed waveforms into subbands, to interpolate and decompress at least one compressed subband, and to synthesize the decompressed subbands as an approximation of the original waveform (e.g., subject to reconstruction error requirements).

A method of operation of a device includes receiving an input signal at the device. The input signal is generated using at least one microphone. The input signal includes a first signal component having a first amount of wind turbulence noise and a second signal component having a second amount of wind turbulence noise that is greater than the first amount of wind turbulence noise. The method further includes generating, based on the input signal, an output signal at the device. The output signal includes the first signal component and a third signal component that replaces the second signal component. A first frequency response of the input signal corresponds to a second frequency response of the output signal.

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.

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.

A noise suppression device includes: an adaptive filter unit that suppresses, using an adaptive filter, a noise component contained in a voice signal generated from a voice captured by a voice input unit to generate a corrected voice signal; a noise generation detection unit that detects timing of generation of the noise component in the voice signal; and a period suppression unit that suppresses the corrected voice signal during a predetermined period of time after the timing of the generation of the noise component.

Systems and methods for controlling adaptivity of noise cancellation are presented. One or more audio signals are received by one or more corresponding microphones. The one or more signals may be decomposed into frequency sub-bands. Noise cancellation consistent with identified adaptation constraints is performed on the one or more audio signals. The one or more audio signals may then be reconstructed from the frequency sub-bands and outputted via an output device.

Systems and methods for controlling adaptivity of noise cancellation are presented. One or more audio signals are received by one or more corresponding microphones. The one or more signals may be decomposed into frequency sub-bands. Noise cancellation consistent with identified adaptation constraints is performed on the one or more audio signals. The one or more audio signals may then be reconstructed from the frequency sub-bands and outputted via an output device.

The present invention relates to a method for receiving data transmitted acoustically. The method includes receiving an acoustically transmitted signal encoding data; processing the received signal to minimise environmental interference within the received signal: and decoding the processed signal to extract the data. The data encoded within the signal using a sequence of tones. A method for encoding data for acoustic transmission is also disclosed. This method includes encoding data into an audio signal using a sequence of tones. The audio signal in this method is configured to minimise environmental interference. A system and software are also disclosed.

The present invention relates to a method for receiving data transmitted acoustically. The method includes receiving an acoustically transmitted signal encoding data; processing the received signal to minimise environmental interference within the received signal: and decoding the processed signal to extract the data. The data encoded within the signal using a sequence of tones. A method for encoding data for acoustic transmission is also disclosed. This method includes encoding data into an audio signal using a sequence of tones. The audio signal in this method is configured to minimise environmental interference. A system and software are also disclosed.

A speech recognition system that automatically sets the volume of output audio based on a sound intensity of a command spoken by a user to adjust the output volume. The system can compensate for variation in the intensity of the captured speech command based on the distance between the speaker and the audio capture device, the pitch of the spoken command and the acoustic profile of the system, and the relative intensity of ambient noise.