Dynamic directional characteristics track sound sources to maintain voice clarity and spatial perception during user movement.
A binaural beat controller adjusts volume levels in each ear to ensure equal sound magnitude.
Dual microphones determine user position via time-of-flight analysis to prevent replay attacks during voice authentication.
Audio processor dynamically allocates loudspeaker signals based on listener position, resolving sweet spot limitations by rendering sound that follows the user.
Segmented acoustic zones deliver personalized audio to vehicle occupants, preventing unwanted sound exposure from other passengers.
Electronic control replaces mechanical dosers to enable dynamic lubricant volume adjustment, eliminating manual reconfiguration.
Direction of arrival information guides nullformer generation, reducing calculation overhead while maintaining extraction accuracy.
Host speaker broadcasts audio data to satellite speakers, extending transmission distance beyond standard Bluetooth limits.
Machine learning classifies sound sources to suppress noise and reverberation, resolving acoustic feedback in arena environments.
Signal processing module emphasizes impulsive sounds across mismatched stereo speakers to optimize audio output.
High-frequency spectral envelope warping translates audio signals to improve speech intelligibility in hearing assistance devices.
A loudspeaker monitoring system uses cross-correlation to assess individual speaker status during playback.
Time-multiplexed beamformers localize multiple sound sources using two microphones, resolving the conflict between device size and localization precision.
A speech providing device detects user position and orientation to deliver corresponding audio cues.
A playback device detects availability to communicate and re-joins a bonded zone without user intervention.
Interactive speaker grille segments faceplate regions to align circuit board openings with acoustic paths for direct user interaction sensing.
A point audio source estimator calculates time frequency tile difference measures between beamformed audio and noise reference signals.
Beamforming processing in a distributed lighting network separates speech sources from ambient noise interference.
Adaptive microphone system limits adaptation parameter ranges based on ambient noise floors to mask non-stationary microphone noise artifacts.
Motion sensors identify user movement time instances to temporarily mute distracting sounds in active hear-through wearable audio devices.
Electronic devices transmit audio data using distinct frequency ranges for identification and payload signals.
A method separates audio signals into frequency bands to detect feedback presence.
A receiver control method detects blockages via impedance changes and adjusts signal gain to reduce noise.
Computer-based optimization adjusts speaker splay angles and positions to achieve target sound pressure levels across a venue.
A directional speech separation system isolates audio sources using time delay analysis and cross-correlation to generate frequency masks.
A virtual broadside scan identifies acoustic sources using directed beams to capture speech signals from multiple talkers.
Adaptive low-pass filtering adjusts cut-off frequencies based on signal estimates to reduce musical noise while maintaining speech clarity.
Multi-band gain control limits cone excursion and thermal overload by applying selective attenuation to frequency bands contributing to mechanical stress.
Wearable ear units deliver directional vibrations to users.
Speech detection directs beamforming toward a user, improving signal-to-noise ratio in noisy environments.
Wearable audio systems determine unique acoustic transfer functions from head geometry to authenticate users without passwords.
Drone acoustic localization splits wideband signals into narrow sub-bands and fuses measurements to overcome propeller ego-noise interference.
Determining a dominant sound source direction selects two microphones to create focused audio signals, reducing bit requirements for transmission.
A binaural hearing aid system shifts masker signals to distinct frequency bands using digital transposition.
An adaptive cue filter processes in-the-ear and behind-the-ear microphone signals to preserve spatial cues while minimizing acoustic feedback.
An audio signal processing system generates personalized hearing assistance policies through automated testing interfaces.
A hearing device filter bank uses separate frequency-dependent and independent attenuation components to assemble partial band signals.
An earphone system directs ambient sound to the receiver using signal processing controlled by user voice activity.
Assigning Virtual Loudspeaker Objects to microphone setups enables interactive playback without explicit room property knowledge.
An object-oriented mapper assigns audio objects to input channels based on temporal overlap.
A head-mounted device adjusts speaker amplitude and phase using a monitoring microphone to reduce sound leakage.
An audio device applies distractor attenuation parameters to microphone signals.
A pressure sensing transducer detects blocked acoustic ports, allowing the system to alter audio characteristics and conserve battery power.
A hearing device system provides user-selectable ambient sound attenuation settings during audio streaming sessions.
Separating the HRTF reconstruction function from bulky hearing devices via an intermediary manifold resolves positional sound data accuracy trade-offs.
Selective amplification of spectral components between 800 Hz and 11,000 Hz improves word recognition by 20% in noisy environments.
A wearable audio output device alters an audio stream to provide mute status indication without visual indicators.
A control unit transmits encoded audio signals via a data network to exercise machines in training classes.
A terminal plays test signals and collects ambient sound to enable server-based acoustic parameter recognition for audio playback device calibration.