A method includes, while operating an audio processing device in a use mode, retrieving first direction of arrival (DOA) data corresponding to a first audio output device from a memory of the audio processing device and generating a first null beam directed toward the first audio output device based on the first DOA data. The method also includes retrieving second DOA data corresponding to a second audio output device from the memory of the audio processing device and generating a second null beam directed toward the second audio output device based on the second DOA data. The first DOA data and the second DOA data are stored in the memory during operation of the audio processing device in a calibration mode.

Audio signal processing device calibration

A voice-activated computing device configured to transmit a pilot tone and then capture or receive a signal, which corresponds to the pilot tone, reflected from within the environment containing the voice-activated computing device. The voice-activated computing device, or some other computing system or device, analyzes the received signal in order to determine analyze one or more characteristics present within the signal, i.e. noise, echo, etc. Based upon the analysis, models for signal processing can be determined, selected and/or altered. Future signals received by the voice-activated computing device can be processed with such models. The analysis can also allow for the models to be dynamically updated and for models to be dynamically created.

Characterizing environment using ultrasound pilot tones

A method for echo reduction by an electronic device is described. The method includes nulling at least one speaker. The method also includes mixing a set of runtime audio signals based on a set of acoustic paths to determine a reference signal. The method also includes receiving at least one composite audio signal that is based on the set of runtime audio signals. The method further includes reducing echo in the at least one composite audio signal based on the reference signal.

Systems and methods for surround sound echo reduction

In one embodiment, a method comprising receiving at a microphone located at a first location audio received from plural speakers, the audio received at a first amplitude level; and responsive to moving the microphone away from the first location to a second location, causing adjustment of the audio provided by the plural speakers to target the first amplitude level at the microphone.

Loudspeaker beamforming for personal audio focal points

Methods and systems specifically designed to reduce loudspeaker distortion by reducing voice coil excursion are provided An input audio signal is processed based on a specific linear model of a loudspeaker, and a dynamic filter is generated and applied to this audio signal The filter changes the relative phases of the spectral components of the input signal to reduce estimated excursion peaks The quality of the audio signal is not diminished in comparison to traditional filter approaches The processing of the input signal may involve determining a main frequency which may be used to determine the fundamental frequency Multiples of the fundamental frequencies provide harmonic frequencies The phases of all the harmonic frequencies, including the fundamental, may be measured and compared to a target vector of phases The difference between the measured phases and the target phases may then be used to calculate the poles and zeros and corresponding filter coefficients

Speaker distortion reduction

A device for suppressing ambient sounds from speech received by a microphone array is provided. One embodiment of the device comprises a microphone array, a processor, an analog-to-digital converter, and memory comprising instructions stored therein that are executable by the processor. The instructions stored in the memory are configured to receive a plurality of digital sound signals, each digital sound signal based on an analog sound signal originating at the microphone array, receive a multi-channel speaker signal, generate a monophonic approximation signal of the multi-channel speaker signal, apply a linear acoustic echo canceller to suppress a first ambient sound portion of each digital sound signal, generate a combined directionally-adaptive sound signal from a combination of each digital sound signal by a combination of time-invariant and adaptive beamforming techniques, and apply one or more nonlinear noise suppression techniques to suppress a second ambient sound portion of the combined directionally-adaptive sound signal.

Ni Xudong

Papers

Adaptive ambient sound suppression and speech tracking

Adaptive ambient sound suppression and speech tracking method and system