Patent classifications
H04R2410/01
Adaptive mixing of sub-band signals
A method includes receiving a first microphone array processing signal (associated with a frequency band that includes a plurality of sub-bands) and receiving a second microphone array processing signal associated with the frequency band. The method includes generating a first output corresponding to a first sub-band based on the first microphone array processing signal and generating a second output corresponding to the first sub-band based on the second microphone array processing signal. The method includes generating a third output corresponding to a second sub-band based on the first microphone array processing signal and generating a fourth output corresponding to the second sub-band based on the second microphone array processing signal. The method includes performing a first set of microphone mixing operations to generate an adaptive mixer output for the first sub-band and performing a different set of microphone mixing operations to generate another adaptive mixer output for the second sub-band.
AUDIO INTERFERENCE CANCELLATION
Methods and systems for audio interference cancellation are disclosed. A first beamforming zone associated with a location of a first audio source may be determined. A second beamforming zone associated with a location of a second audio source may be determined. Based on determining that first audio associated with the first audio source dominates a first frequency band associated with the first beamforming zone and the second beamforming zone, prevention of attenuation of audio output in the first beamforming zone and within the first frequency band may be caused. Based on determining that second audio associated with the second audio source dominates a second frequency band associated with the first beamforming zone and the second beamforming zone, attenuation of audio output in the first beamforming zone and within the second frequency band may be caused.
Microphone array with automated adaptive beam tracking
An example method of operation may include designating sub-regions which collectively provide a defined reception space, receiving audio signals at a controller from the microphone arrays in the defined reception space, configuring the controller with known locations of each of the microphone arrays, assigning each of the sub-regions to at least one of the microphone arrays based on the known locations, and creating beamform tracking configurations for each of the microphone arrays based on their assigned sub-regions.
DYNAMIC BEAMFORMING TO IMPROVE SIGNAL-TO-NOISE RATIO OF SIGNALS CAPTURED USING A HEAD-WEARABLE APPARATUS
Method to perform dynamic beamforming to reduce SNR in signals captured by head-wearable apparatus starts with microphones generating acoustic signals. Microphones are coupled to first stem of the apparatus and to second stem of the apparatus. First and second beamformers generate first and second beamformer signals, respectively. Noise suppressor attenuates noise content from the first beamformer signal and the second beamformer signal. Noise content from first beamformer signal are acoustic signals not collocated in second beamformer signal and noise content from second beamformer signal are acoustic signals not collocated in first beamformer signal. Speech enhancer generates clean signal comprising speech content from first noise-suppressed signal and second noise-suppressed signal. Speech content are acoustic signals collocated in first beamformer signal and second beamformer signal.
Soundfield decomposition, reverberation reduction, and audio mixing of sub-soundfields at a video conference endpoint
At a microphone array, a soundfield is detected to produce a set of microphone signals each from a corresponding microphone in the microphone array. The set of microphone signals represents the soundfield. The detected soundfield is decomposed into a set of sub-soundfield signals based on the set of microphone signals. Each sub-soundfield signal is processed, such that each sub-soundfield signal is separately dereverberated to remove reverberation therefrom, to produce a set of processed sub-soundfield signals. The set of processed sub-sound field signals are mixed into a mixed output signal.
Generating an audio signal from multiple microphones based on uncorrelated noise detection
An audio capture device selects between multiple microphones to generate an output audio signal depending on detected conditions. When the presence of wind noise or other uncorrelated noise is detected, the audio capture device selects, for each of a plurality of different frequency sub-bands, an audio signal having the lowest noise and combines the selected frequency sub-bands signals to generate an output audio signal. When wind noise or other uncorrelated noise is not detected, the audio capture device determines whether each of a plurality of microphones are wet or dry and selects one or more audio signals from the microphones depending on their respective conditions.
Sensor management for wireless devices
A system and method for selecting audio capture sensors of wearable devices in obtaining voice data. The method provides obtaining signals associated with the user's voice at first and second wearable devices, comparing energy levels of the first and second signals, and selecting one or more audio capture sensors based on the energy levels of each signal. Due to the symmetry of the acoustic energy produced by the user's voice to a first and second wearable device, any difference in energy level between the total energy obtained by the first wearable device and the total energy obtained by the second wearable device can be attributed solely to ambient noise. Thus, the device with the higher total energy has a lower signal-to-noise ratio and selection of an audio capture sensor of the other wearable device with a higher signal-to-noise ratio is provided to obtain voice data moving forward.
Optimization of multi-microphone system for endpoint device
In one embodiment, a multi-microphone system for an endpoint device receives input signals for a remote conference between the endpoint device and at least one other endpoint device. The multi-microphone system may include at least a top microphone unit and a bottom microphone unit. A signal degradation event that causes degradation of signals received by the top microphone unit or the bottom microphone unit is detected. Then, based on information regarding the signal degradation event, it is determined whether the signal degradation event affects one or both of the top microphone unit and the bottom microphone unit. In response, an output signal is generated for transmission to the at least one other endpoint device, and the output signal uses a portion of the input signals that excludes signals received by the top microphone unit and/or the bottom microphone unit determined to be affected by the signal degradation event.
SOUND PICK-UP APPARATUS AND METHOD
To improve, when area sound pick-up is performed to collect sounds from a sound source in a target area, the sound quality of the collected sounds. The present invention relates to a sound pick-up apparatus that performs area sound pick-up. The sound pick-up apparatus calculates a sound volume level of a mixing signal to mix with a target area sound on the basis of power of estimated noise obtained by estimating background noise included in an input signal input from a microphone array, and power of a non-target area sound, adjusts a sound volume level of the input signal, and a sound volume level of the estimated noise to mix with the mixing signal on the basis of the sound volume level of the calculated mixing signal, and generates and outputs a mixed target area sound with which the input signal that is adjusted to have the calculated sound volume level and the estimated noise that is adjusted to have the calculated sound volume level are mixed.
Sound pickup device, program recorded medium, and method
A sound pickup device is provided, the device including (1) a directionality forming unit that forms directionality to output of a microphone array, (2) a target area sound extraction unit that extracts non-target area sound from output of the directionality forming unit, and that suppresses non-target area sound components extracted from output of the directionality forming unit so as to extract target area sound, (3) a determination information computation unit that computes determination information, (4) an area sound determination unit that determines whether or not target area sound is present using the determination information computed by the determination information computation unit, and (5) an output unit that outputs the target area sound extracted only in cases in which the target area sound is determined to be present by the area sound determination unit.