Embodiments allow for an auto-mixer to gate microphones on and off based on speech detection, without losing or discarding the speech received during the speech recognition period. An example method includes receiving and storing an input audio signal. The method also includes determining, based on a first segment of the input audio signal, that the input audio signal comprises speech, and determining a delay between the input audio signal and a corresponding output audio signal provided to a speaker. The method also includes reducing the delay, wherein reducing the delay comprises removing one or more segments of the stored input audio signal to create a time-compressed audio signal and providing the time-compressed audio signal as the corresponding output audio signal. The method also includes determining that the delay is less than a threshold duration, and responsively providing the input audio signal as the corresponding output audio signal.

A system and method for processing an audio input signal includes a microphone, a controller, and a communication link that may be coupled to a remote speaker. The microphone captures the audio input signal and communicates the audio input signal to the controller, and the controller is coupled to the communication link. The controller includes executable code to generate, via a linear noise reduction filtering algorithm, a first resultant based upon the audio input signal, and generate, via non-linear post filtering algorithm, a second resultant based upon the first resultant. An audio output signal is generated based upon the second resultant employing a feature restoration algorithm. The audio output signal is communicated, via the communication link, to a speaker that may be at a remote location.

Various implementations include systems for processing audio signals. In particular implementations, a system for processing audio signals includes: an accessory device that includes a first processor for running a machine learning model on an input signal, where the machine learning model includes a classifier configured to generate metadata associated with the input signal; and a wearable audio device configured for wireless communication with the accessory device, the wearable audio device including a second processor that utilizes the metadata from the accessory device to process a source audio signal and output a processed audio signal.

Disclosed is an operation device that executes a noise removal process of removing noise from a collected audio signal collected by a microphone. In an acting state where noise generation possibly occurs, the operation device transmits noise generation information indicating that the operation device is in the acting state where noise generation possibly occurs, and changes the details of the noise removal process, according to the noise generation information.

The present application relates to a system and method for providing autoregressive based residual echo suppression in the STFT domain in a bidirectional vehicle communications system including receiving, from a communications processor, a speaker signal for coupling to a speaker, receiving, from a microphone, a microphone signal wherein the microphone signal includes a voice signal and a residual echo signal, transforming the signals to the STFT domain generating an estimated power spectral density of the residual echo signal in response to a prior power spectral density of a prior residual echo signal, isolating the voice signal from the microphone signal by multiplying the estimated residual echo gain generated using the estimated power spectral density of the residual echo signal with the microphone signal, transforming the signals back to the time domain and coupling the voice signal to the communications processor.

A method for preparing reference signals for an echo cancellation system disposed in a vehicle, comprising the steps of: receiving a plurality of drive signals, each drive signal being provided to an associated transducer of a plurality of acoustic transducers such that the associated acoustic transducer transduces the drive signal into an acoustic signal, filtering each drive signal with a respective filter of a plurality of filters to produce a plurality of filtered signals, wherein each of the plurality of filters approximates a transfer function from an associated acoustic transducer to a microphone disposed within the vehicle such that the plurality of filtered signals each estimate a respective acoustic signal at the microphone; summing together at least a subset of the plurality of filtered signals to produce a summed reference signal; and outputting the summed reference signal to an echo cancellation system.

A method and a device for updating a coefficient vector of a finite impulse response filter are provided. The update method includes: obtaining an updated step-size diagonal matrix for a coefficient vector of the FIR filter; and obtaining an updated coefficient vector of the FIR filter based on the updated step-size diagonal matrix.

A system comprises an ear-worn electronic device configured to be worn by a wearer. The ear-worn electronic device comprises a processor and memory coupled to the processor. The memory is configured to store an annoying sound dictionary representative of a plurality of annoying sounds pre-identified by the wearer. A microphone is coupled to the processor and configured to monitor an acoustic environment of the wearer. A speaker or a receiver is coupled to the processor. The processor is configured to identify different background noises present in the acoustic environment, determine which of the background noises correspond to one or more of the plurality of annoying sounds, and attenuate the one or more annoying sounds in an output signal provided to the speaker or receiver.

The present application discloses a method for determining a delay between signals, an apparatus, a device and a storage medium, and relates to voice technology. In the method, apparatus, device, and storage medium provided by the present disclosure, by performing down-sampling processing on the signals, the amount of calculation for determining the delay can be reduced, thereby improving the determination efficiency. Moreover, signal segments including alignment positions of the two signals can be estimated in the signal through a currently determined delay, and then the processing can be performed again on the signal segments. In this way, a range for determination can be gradually reduced, that is, an accurate delay can be obtained by just processing shorter signals, which not only ensures the accuracy of the determination, but also reduces the amount of data processing.

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.