Abstract: The present disclosure provides an audio playback system adapted to automatically switch an active microphone back and forth between two or more devices. For example, where the system is a pair of earbuds, where each earbud is worn by a separate user, the system may switch the active microphone to the device worn by the user that is speaking at a given time. While that device holds the active microphone, the other device may wait until a particular event that frees up the microphone, such as if the user wearing the device with the active microphone stops talking. According to some examples, a notification may be provided through one or more of the devices in the system to let the user know, for example, that he does not have the active microphone, that the active microphone is free, that the active microphone has been switched, etc.
Abstract: User equipment (UE) includes a housing, an audio output module, and a drive module connected to the audio output module. The drive module is configured to drive the audio output module to move along a straight line between a first location and a second location. The first location is inside the housing. The second location is outside the housing. When the audio output module is in an audio output state, the drive module is configured to drive the audio output module to move along the straight line to the second location. When the audio output module exits the audio output state, the drive module is configured to drive the audio output module to move along the straight line to the first location.
Type:
Grant
Filed:
March 13, 2020
Date of Patent:
June 14, 2022
Assignee:
Beijing Xiaomi Mobile Software Co., Ltd.
Abstract: An abnormal noise determination apparatus including a microphone array disposed inside a vehicle and an electronic control unit including a microprocessor. The microprocessor is configured to perform acquiring an abnormal noise data on an abnormal noise generated by a sound source disposed in a predetermined position inside the vehicle, the abnormal noise data including an information on a strength and a generation direction of the abnormal noise collected by the microphone array in advance or assumed to be collected by the microphone array; acquiring a traveling noise data including an information on a strength and a generation direction of a traveling noise collected by the microphone array during traveling of the vehicle; and determining whether the abnormal noise is included in the traveling noise of the vehicle, based on the abnormal noise data acquired and the traveling noise data acquired.
Abstract: A method and system for identifying a sensor node located closest to the origin of an audio signal. There can be at least three sensor nodes connected to a computational node, and each sensor node includes an audio directional sensor and a device for providing a reference direction. The sensor nodes can receive the audio signal and each audio directional sensor can provide an angle of propagation of the audio signal relative to the reference direction. The angular mean of the measured angles of propagation from all sensor nodes is calculated and the sensor node providing the angle which is closest to the angular mean is defined as the sensor node being closest to the origin of the audio signal.
Abstract: Example techniques relate to device spaces and default designations in a media playback system. A device space may create an association between a networked microphone device and one or more playback devices such that certain voice commands (e.g., playback commands) received by the networked microphone device are used to control the one or more playback devices (unless otherwise designated in the voice command). Furthermore, in bonded pairs and bonded groups of playback devices that include at least one NMD, certain playback devices within the bonded pair or group may be designated as default so as to avoid multiple responses to a voice input.
Abstract: An apparatus for decoding an encoded multichannel signal includes: a base channel decoder for decoding an encoded base channel to obtain a decoded base channel; a decorrelation filter for filtering at least a portion of the decoded base channel to obtain a filling signal; and a multichannel processor for performing a multichannel processing using a spectral representation of the decoded base channel and a spectral representation of the filling signal, wherein the decorrelation filter is a broad band filter and the multichannel processor is configured to apply a narrow band processing to the spectral representation of the decoded base channel and the spectral representation of the filling signal.
Type:
Grant
Filed:
January 9, 2020
Date of Patent:
May 24, 2022
Assignee:
FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V.
Inventors:
Jan Büthe, Franz Reutelhuber, Sascha Disch, Guillaume Fuchs, Markus Multrus, Ralf Geiger
Abstract: A signal processing apparatus includes: an audio signal processing unit configured to perform wavefront synthesis processing for at least part of a plurality of sound source data; a first output unit configured to output N-channel audio signals output from the audio signal processing unit to a first speaker device; a mix processing unit configured to mix the N-channel audio signals output from the audio signal processing unit; and a second output unit configured to output an audio signal output from the mix processing unit to a second speaker device.
Abstract: A dual microphone signal processing arrangement for reducing reverberation is described. Time domain microphone signals are developed from a pair of sensing microphones. These are converted to the time-frequency domain to produce complex value spectra signals. A binary gain function applies frequency-specific energy ratios between the spectra signals to produce transformed spectra signals. A sigmoid gain function based on an inter-microphone coherence value between the transformed spectra signals is applied to the transformed spectra signals to produce coherence adapted spectra signals. And an inverse time-frequency transformation is applied to the coherence adjusted spectra signals to produce time-domain reverberation-compensated microphone signals with reduced reverberation components.
Abstract: A method of inspecting a sound input/output device is disclosed. A method of inspecting a sound input/output device according to an embodiment of the present disclosure can diagnose an error state of either a speaker or a microphone based on a cross-correlation of input/output signals by receiving a sound signal from an AI device through the microphone. The method of inspecting of the present disclosure may be associated with an artificial intelligence module, a drone ((Unmanned Aerial Vehicle, UAV), a robot, an AR (Augmented Reality) device, a VR (Virtual Reality) device, a device associated with 5G services, etc.
Abstract: A sound source separation system includes: a controller that: acquires pieces of sound collection data with microphones that collect sounds output from first and second sound sources. The first sound source is at a first position at which effective distances from the microphones are equal and the second sound source is at a different position. The controller further acquires, based on the sound collection data, frequency spectra in two dimensions of a circumferential direction of a circle and a time direction. The first position is a center of the circle and each of the effective distances is a radius of the circle. The controller further separates, from the frequency spectra, a first sound source spectrum and a second sound source spectrum.
Abstract: According to an embodiment, an electronic device may include a memory configured to store a noise removal neural network model and data utilized in the noise removal neural network model and a processor electrically connected to the memory wherein the memory may store instructions that, when executed, enable the processor to: output a first channel signal using a first beamformer for a multi-channel audio signal; output a second channel signal using a second beamformer; generate a third channel signal that compensates for a difference in noise levels between the first channel signal and the second channel signal; and train the noise removal neural network model by using the third channel signal in which the difference in noise levels is compensated for and the first channel signal as input values.
Abstract: Disclosed herein are techniques for determining a wearing position of a boomless headset. An earpiece of the boomless headset can include at least one local talker (LT) microphone and a reference microphone. The LT microphone(s) are disposed substantially in a first end of the earpiece closest to a mouth of a LT when the LT wears the earpiece. The reference microphone is disposed substantially in a second end of the earpiece, furthest from the mouth of the LT when the LT wears the earpiece. A signal strength measurement (SSM) for a local talker audio signal to the LT microphone(s) and a SSM for a signal to the reference microphone are obtained. Signal processing logic can determine whether the earpiece is worn at an incorrect ear based on whether a difference between the SSM for the LT microphone(s) and the SSM for the reference microphone is below a predetermined threshold.
Abstract: A method, apparatus and computer program product are provided for vehicle localization via frequency audio features. In this regard, a frequency audio signature of audio data received from one or more audio sensors of a vehicle is determined. Location data associated with the vehicle is also determined. Based on the location data for the vehicle, at least a portion of an audio feature map is selected. The audio feature map stores frequency audio signatures associated with road noise in relation to respective locations. Furthermore, the frequency audio signature of the audio data is compared with the frequency audio signatures of the audio feature map. Based on the comparison between the frequency audio signature of the audio data and the frequency audio signatures of the audio feature map, a location of the vehicle is refined to generate updated location data for the vehicle.
Abstract: A sound speaker for a vehicle is provided. The sound speaker includes a flat surface with a rod extending perpendicular from the surface, a cylindrical electro-dynamic exciter with an axial longitudinal through hole arranged for receiving the rod and a circular clamp for securing the electro-dynamic exciter on the first flat surface in a closed position. The electro-dynamic exciter includes a longitudinal rib inside the through hole and the rod includes a longitudinal recess for receiving the rib. The rod further includes a lateral recess for receiving the clamp. A least a part of the lateral recess forms a conical frustum and the clamp in the closed position forms a corresponding conical frustum. This part of the lateral recess pushes the clamp in the closed position against the electro-dynamic exciter towards the flat surface.
Type:
Grant
Filed:
June 26, 2020
Date of Patent:
April 5, 2022
Inventors:
Eduard Cornel Petca, Jens Friedrich, Robert Joest, Johannes Kerkmann, Costinel Alexandru Ionica, Dimitrios Patsouras, Stephan Eisele
Abstract: Techniques are described for detecting and correcting mismatched microphone sensitivities in a microphone array without knowledge of the acoustic excitation source(s). Mismatch is detectable based on time and/or frequency domain analysis of each microphone's long term exposure to a real-world sound field that includes an acoustic source and a non-acoustic source, and is corrected by adjusting the amount of amplification applied to at least one microphone signal. In the time domain, sensitivity matching can be performed by using an average of all microphone signals as a reference signal. In some embodiments, the reference signal is the root mean square of the average. Alternatively, a single microphone can be selected as a reference. In some embodiments, sensitivity mismatch is detected and corrected at specific frequencies based on comparing frequency components of amplified microphone signals. Sensitivity matching can be repeated to ensure that the microphones remain sensitivity-matched over time.
Abstract: The present technology relates to a headphone and a reproduction control method that enable external sound to be caught more easily and promptly, and a program. The headphone includes: a sound collecting unit configured to collect outer sound; a detection unit configured to detect a specific motion to a sensor unit for capture of the outer sound; and a reproduction control unit configured to cause, in a case where the specific motion is detected, the outer sound collected by the sound collecting unit to be reproduced and volume of audio under reproduction to be reduced or the reproduction of the audio to stop. The present technology can be applied to, for example, a headphone.
Abstract: A specialized audio/instrument cable adapter with built-in digital signal processing capabilities that adds user-defined audio effects (such as reverb, delay, chorus and/or distortion) from within the cable adapter itself to affect the sound generated from an instrument or microphone such that the cable adapter (with a instrument/microphone cable) is the only connection needed between the instrument or microphone and an output device (such as an amplifier, PA, powered speaker, music mixer, or a recording device). The audio effects used by the cable adapter can be changed via (i) an app from a smartphone, tablet, computer or other electronic device; (ii) a wireless controller that attaches to the instrument; (iii) a pedal, and/or (iv) any other type of wireless controller that has the ability to communicate with a smartphone/tablet/computer or other electronic device.
Type:
Grant
Filed:
June 22, 2021
Date of Patent:
March 29, 2022
Inventors:
Bobby Elijah Aviv, David G. Aviv, Mark Schaffel, Guy Zohar
Abstract: A method for generating an audio signal representing the dynamic state of a land vehicle as a function of the value of at least one first parameter comprises the steps of receiving an input audio signal and dividing the input audio signal into a set of audio frames of given duration; selecting, an audio frame representing the input audio signal; subdividing the selected audio frame into a set of audio sub-frames; determining a probability of transition between orderly pairs of the audio sub-frames and synthesizing the audio signal representing the dynamic state of a land vehicle by generating a sequence of audio sub-frames; computing a centre frequency of the selected audio frame and carrying out a frequency shift of the centre frequency as a function of the centre frequency of the selected audio frame and of the value of at least one first parameter that indicates the dynamic state of the vehicle, thereby generating a frequency-shifted subsequent audio sub-frame.
Type:
Grant
Filed:
July 11, 2019
Date of Patent:
March 22, 2022
Assignee:
MARELLI EUROPE S.p.A.
Inventors:
Simona De Cesaris, Saverio Armeni, Walter Nesci
Abstract: One or more embodiments include techniques for generating audio for a speaker system worn by a listener. The speaker system analyzes an audio input signal to determine that a sound component of the audio input signal has an apparent location that is at a vertical distance from a listener. The speaker system selects an externally facing speaker included in the speaker system that faces at least partially upward or at least partially downward based on the vertical distance from the listener. The speaker system transmits the sound component to the externally facing speaker.
Type:
Grant
Filed:
July 6, 2020
Date of Patent:
March 22, 2022
Assignee:
Harman International Industries, Incorporated