Headphones provide binaural sound to a user wearing the headphones. The headphones store HRTFs of the user and include head tracking and a digital signal processor (DSP). The DSP processes sound into binaural sound that externally localizes to a sound localization point (SLP) at least one meter away from a head of the user. The headphones receive head gestures from the user.

A method provides a voice in binaural sound to a user. The method includes designating a sound localization point (SLP) with a handheld portable electronic device (HPED) and tracking the SLP with electronic glasses worn on a head of the user. A processor processes the voice with head-related transfer functions (HRTFs) to generate the voice in the binaural sound that localizes at the SLP at a location in empty space at physical object.

A signal processing method includes: multiplying at least one of M signals output from M microphones by a gain so as to equalize sound pressure levels of the M signals, the M signals representing sounds that arrive at the M microphones from a sound source located within a predetermined distance from the M microphones, M being an integer equal to or greater than two; delaying at least one of the M signals so as to resolve time discrepancies between the M signals, the time discrepancies being caused by differences in arrival time between sounds that arrive at the M microphones from the sound source; and applying a filter so as to suppress a signal which is included in the M signals obtained through the multiplying and the delaying and which represents a sound output from the sound source located within the predetermined distance.

A two-way conversation assisting device includes a first microphone that enters a first voice, a first loudspeaker that outputs the first voice, a second microphone that enters a second voice, a second loudspeaker that outputs the second voice, and a first echo and crosstalk canceller. The first echo and crosstalk canceller estimates and calculates, using an input signal into the second loudspeaker, a first interference signal indicative of degrees of a first echo caused when the second voice output from the second loudspeaker enters into the first microphone and first crosstalk caused when the second voice enters into the first microphone, and removes the calculated first interference signal from an output signal of the first microphone.

Systems and methods for acoustic feedback cancellation. One method includes transmitting, with a transceiver, a known full-band sequence at a first time. The method includes receiving, via a microphone, a received audio signal including a received copy of the known full-band sequence. The method includes filtering the received audio signal to generate a filtered audio signal, and detecting, based on the filtered audio signal, a second portable communication device in proximity to the portable communication device. The method includes, in response to detecting the second portable communication device, determining, based on the first time and filtered audio signal, an estimated loop delay. The method includes initializing an adaptive feedback cancellation filter based on the estimated loop delay and the plurality of estimated filter coefficients. The method includes determining a plurality of estimated filter coefficients based on the known full-band sequence and the received copy of the known full-band sequence.

A microphone picks-up voice of a driver. A first echo suppression unit outputs a voice signal after first echo suppression based on a voice signal of the driver and a voice signal after echo suppression in the past (first reference signal) stored in a buffer memory. A second echo suppression unit outputs a voice signal after second echo suppression based on a voice signal of the driver and a voice signal after the echo suppression in the past (second reference signal) stored in a buffer memory. An output signal selector selects one of the voice signals after the first echo suppression or the voice signal after the second echo suppression according to a detection result of the presence or absence of a system variation by a system variation detector, and causes a speaker to output the selected voice signal.

Provided is a call quality improvement method configured to operate a call quality improvement system and a call quality improvement apparatus by executing an artificial intelligence (AI) algorithm and/or a machine learning algorithm in a 5G environment connected for the Internet of Things. According to one embodiment of the present disclosure, the call quality improvement method may include receiving a voice signal from a far-end speaker, receiving a sound signal including a voice signal from a near-end speaker, receiving an image of a face of the near-end speaker, including lips, and extracting the voice signal of the near-end speaker from the received sound signal.

A method for detecting and flagging problems in videoconferencing is provided. The method includes collecting, by a processing device of a user videoconferencing device, one or more sets of acoustic parameters associated with a conference room. Each set of the one or more sets of acoustic parameters is associated with a video conference attended by a corresponding plurality of users. The method further includes determining whether a first set of the one or more sets of acoustic parameters suggest an action to be performed to improve the first set of acoustic parameters associated with the video conference. The method further includes providing at least one of the one or more sets of acoustic parameters associated with the conference room or the action to be performed to improve the first set of acoustic parameters.

During an electronic call between two individuals, a sound localization point simulates a location in empty space from where an origin of a voice of one individual occurs for the other individual.

An intercommunication system with Adaptive Transmit Delay patches audio sources to a radio system (e.g., a trunked-radio system) such that outgoing audio is sent to the radio system as soon as a channel has been acquired, without further transmit delay. The intercommunication system generates a pattern representative of the radio's Talk Permit Tone. The intercommunication system then buffers outgoing audio while analyzing incoming audio from the radio and comparing the incoming audio to the generated pattern to determine when a Talk Permit Tone has been received. When the Talk Permit Tone is received, the intercommunication system releases buffered outgoing audio for transmission.