Patent classifications
H04R2430/25
Acoustic echo cancellation control
Techniques for improving acoustic echo cancellation are described. Energy levels of audio data received from a microphone and representing near-end audio and reference audio data representing far-end audio are determined. If near-end audio is detected but far-end audio is not detected, a controller turns of or bypasses an acoustic echo cancellation system until far-end audio is again detected, thereby decreasing or eliminating distortion of the near-end audio by the acoustic echo cancellation system.
Autonomously motile device with acoustic echo cancellation
A device capable of motion includes an acoustic echo canceller for cancelling a reference signal from received audio data. The device updates an adaptive filter as the device moves to reflect the changing audio channel between a loudspeaker and a microphone of the device. A step size for changing coefficients of the filter is determined based on its velocity. A number of iterations for updating the filter using a frame of audio data is also determined based on the velocity.
SPATIAL AUDIO ARRAY PROCESSING SYSTEM AND METHOD
A spatial audio processing system operable to enable audio signals to be spatially extracted from, or transmitted to, discrete locations within an acoustic space. Embodiments of the present disclosure enable an array of transducers being installed in an acoustic space to combine their signals via inverting physical and environmental models that are measured, learned, tracked, calculated, or estimated. The models may be combined with a whitening filter to establish a cooperative or non-cooperative information-bearing channel between the array and one or more discrete, targeted physical locations in the acoustic space by applying the inverted models with whitening filter to the received or transmitted acoustical signals. The spatial audio processing system may utilize a model of the combination of direct and indirect reflections in the acoustic space to receive or transmit acoustic information, regardless of ambient noise levels, reverberation, and positioning of physical interferers.
SPATIAL AUDIO ZOOM
In an aspect, a lens is zoomed in to create a zoomed lens. Lens data associated with the lens includes a direction of the lens relative to an object in a field-of-view of the zoomed lens and a magnification of the object resulting from the zoomed lens. An array of microphones capture audio signals including audio produced by the object and interference produced by other objects. The audio signals are processed to identify a directional component associated with the audio produced by the object and three orthogonal components associated with the interference produced by the other objects. Stereo beamforming is used to increase a magnitude of the directional component (relative to the interference) while retaining a binaural nature of the audio signals. The increase in magnitude of the directional component is based on an amount of the magnification provided by the zoomed lens to the object.
Directional acoustic sensor, and methods of adjusting directional characteristics and attenuating acoustic signal in specific direction using the same
Disclosed are a directional acoustic sensor, a method of adjusting directional characteristics using the directional acoustic sensor, and a method of attenuating an acoustic signal in a specific direction using the directional acoustic sensor. The directional acoustic sensor includes a plurality of resonance units arranged to have different directionalities and a signal processor configured to adjust directional characteristics by calculating at least one of a sum of and a difference between outputs of the resonance units. In this state, the signal processor attenuates an acoustic signal in a specific direction by using a plurality of directional characteristics obtained by calculating at least one of the sum of and the difference between the outputs of the resonance units at a certain ratio.
System and method for generating audio featuring spatial representations of sound sources
Systems and methods for spatially emulating a sound source. An apparatus includes a microphone array including microphones; and a sound profiler communicatively connected to the microphone array, the sound profiler including a processing circuitry and a memory which contains instructions that, when executed by the processing circuitry, configure the apparatus to: generate synthesized audio based on sound beam metadata, a sound profile, and target listener location data, wherein the sound beam metadata includes timed sound beams defining a directional dependence of a spatial sound wave, wherein the sound profile includes timed sound coefficients determined based on audio signals captured in a space wherein the target listener location data includes a position and an orientation, wherein the synthesized audio emulates sound that would be heard by a listener at the position and orientation of the target listener location data; and providing the synthesized audio for projection.
INFORMATION PROCESSING DEVICE AND CONTROL METHOD
An information processing device includes a signal acquisition unit that acquires a voice signal of an object person outputted from a mic array and a control unit that acquires at least one of noise level information indicating a noise level of noise and first information as information indicating whether or not an obstructor is speaking while obstructing speech of the object person and changes a beam width as a width of a beam corresponding to an angular range of acquired sound, centering at the beam representing a direction in which voice of the object person is inputted to the mic array, and dead zone formation intensity as a degree of suppressing at least one of the noise and voice of the obstructor inputted to the mic array based on at least one of the noise level information and the first information.
Directional acoustic sensor, and methods of adjusting directional characteristics and attenuating acoustic signal in specific direction using the same
Disclosed are a directional acoustic sensor, a method of adjusting directional characteristics using the directional acoustic sensor, and a method of attenuating an acoustic signal in a specific direction using the directional acoustic sensor. The directional acoustic sensor includes a plurality of resonance units arranged to have different directionalities and a signal processor configured to adjust directional characteristics by calculating at least one of a sum of and a difference between outputs of the resonance units. In this state, the signal processor attenuates an acoustic signal in a specific direction by using a plurality of directional characteristics obtained by calculating at least one of the sum of and the difference between the outputs of the resonance units at a certain ratio.
Situational awareness, communication, and safety for hearing protection devices
An apparatus for hearing protection comprises a pair of earpads, a band, microphones, speakers, vibration generators, and a processing unit. Each of the earpads is placed over an ear of the user. The band extends between the pair of earpads. The microphones convert acoustic signals into electrical signals. The speakers are located on each of the pair of earpads and direct sound towards the ear. The vibration generators are located on at least one of the pair of earpads and generate vibratory feedback to the user. The processing unit is connected to the microphones, the speakers, and the vibration generators, and compares first parameters of the electrical signals from the microphones with second parameters of predetermined warning sounds to determine whether the electrical signals comprise one or more of the predetermined warning sounds. If the processing unit determines that the electrical signals comprise one or more of the predetermined warning sounds, the processing unit transmits a warning to the speakers and the vibration generators.
Pole-zero blocking matrix for low-delay far-field beamforming
A system performs pole-zero or IIR modeling and estimation of an inter-microphone transfer function between first and second microphones that output respective first and second microphone signals. The system includes a first adaptive FIR filter to which the first microphone signal is provided, a delay element that delays the second microphone signal by a predetermined delay amount, and a second adaptive FIR filter to which the delayed second microphone signal is provided. A first coefficient of the second adaptive FIR filter is constrained to a fixed non-zero value. The filters are jointly adapted to minimize an error signal that is a difference of the two filters outputs. The delay is small: approximately the acoustic propagation delay between the two microphones and is not determined by the environmental reverberation characteristics. The error signal may serve as a noise reference in a noise canceller, for implementing far-field beamforming with low delay.