A method in a soundfield-capturing endpoint and the capturing endpoint that comprises a microphone array capturing soundfield, and an input processor pre-processing and performing auditory scene analysis to detect local sound objects and positions, de-clutter the sound objects, and integrate with auxiliary audio signals to form a de-cluttered local auditory scene that has a measure of plausibility and perceptual continuity. The input processor also codes the resulting de-cluttered auditory scene to form coded scene data comprising mono audio and additional scene data to send to others. The endpoint includes an output processor generating signals for a display unit that displays a summary of the de-cluttered local auditory scene and/or a summary of activity in the communication system from received data, the display including a shaped ribbon display element that has an extent with locations on the extent representing locations and other properties of different sound objects.

A wearable device comprising: a band unit; and a vibration generating unit coupled to the band unit. The band unit comprises a communication unit receiving a first signal by a predetermined communication method and a control unit determining an amplification extent of the first signal received from the communication unit and generating a second signal based on the determined amplification extent, and the vibration generating unit receives the second signal from the control unit and generates a vibration corresponding to the second signal.

Methods and systems are disclosed for echo suppression of an input audio signal. A multimedia system and a remote control unit are configured to adaptively and dynamically preform calibration processes for obtaining sets of echo suppression parameters. The sets of echo suppression parameters are used to generate an audio signal from an input audio. When rendered by a speaker, the generated audio signal produces echo-suppressed sound of the input audio signal at one of more locations.

A nearend speech detector for classifying speech at a communication system receiving a microphone signal from a nearend microphone and a farend signal from a farend communication system, the nearend speech detector comprising: a signal processor configured to transform the microphone and farend signals into the frequency domain; a calculation unit configured to form: an estimate of a nearend signal representing nearend speech present in the microphone signal; and a measure of gain between the microphone signal and the nearend signal; and a signal classifier configured to classify speech at the communication system in dependence on a measure of variance of the gain and a measure of variance of the nearend signal.

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.

A communications device having a user configurable speakerphone feature is provided. A user may be provided with a plurality of selectable settings for configuring the operation of a microphone or the processing of speech. For example, a user may select settings to filter ambient noise for a cleaner sound or increase a microphone's sensitivity in a conference call environment. In one embodiment, the user may select a setting that changes a microphone level or microphone operating range. When the microphone detects speech, the speech may be processed in accordance with user selected processing settings. For example, a user may select settings to specify a gain control, a frequency response, or an activity threshold. Thus, a user may configure a speakerphone to improve sound quality on the receiving end.

A smartphone provides a voice of a person during a telephone call to a user of the smartphone in binaural sound. The smartphone stores HRTFs of the user, and a digital signal processor (DSP) processes the voice of the person with the HRTFs so the voice of the person externally localizes in the binaural sound at a sound localization point (SLP) that is in empty space at least one meter away from a head of the user. The smartphone displays an AR image at the SLP during the telephone call.

A smartphone provides a voice of a person during a telephone call to a user of the smartphone in binaural sound. The smartphone stores HRTFs of the user, and a digital signal processor (DSP) processes the voice of the person with the HRTFs so the voice of the person externally localizes in the binaural sound at a sound localization point (SLP) that is in empty space one meter away from a head of the user.

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.

Methods and apparatus include headphones that determine a size of a head of a listener and a shape of an ear of the listener. The size of the head and the shape of the ear determine head-related transfer functions (HRTFs) that are customized to the listener. The headphones include head tracking, and sound is adjusted so that a location of a source of the sound continues to originate from a fixed sound localization point (SLP) in empty space that is at least one meter away from the head of the listener while the head orientations of the listener change with respect to the fixed location.