Signal Processing TRIZ Cases

Aggregating multi-channel audio features via machine learning resolves settling time and robustness issues in sound source localization.

A signal characteristic determinator computes generalized intensity vectors from weighted spherical harmonic coefficients to estimate sound field direction.

A closed-loop system corrects driving signals based on acoustic feedback, ensuring consistent sound output across varying attachment surfaces.

A multi-channel audio processing system segments frequency bands to control emphasis in overlapping sound directions.

An adaptive system determines feedback path parameters using filter coefficients to manage acoustic signals.

Audio tuner applies convergence functions to sampled cabin noise profiles, generating calibrated sound outputs that replicate desired engine dynamics.

Dynamic gain control mitigates environmental overlay instability, allowing external sounds to remain audible while suppressing acoustic interference.

A dual hearing aid system splits audio into low and high tone bands, allowing a controller to adjust gain and prevent consonants from being masked by vowels.

Segmenting the array into sub-arrays directs low-frequency energy to end units, resolving wide beam width and poor distant listening clarity.

Segmenting multichannel playback into a restricted mono reference reduces computational complexity while maintaining speech recognition accuracy.

Audio processing system distinguishes own voice from ambient sound using activity detection and applies distinct gain values to each signal component.

A binaural audio rendering method filters sound programs into low and high frequency portions to create an in-head spatial experience.

Segmented anterior and posterior pendants with neck straps position audio outputs to maintain a stable stereo image regardless of head tilt.

A spectral pop noise removal mask suppresses impulsive components in audio signals.

A hearing instrument detects speech intervals and increases processed signal amplitude based on instantaneous pitch or amplitude derivatives.

A user interface displays noise exposure levels through graphical objects that change color and size based on real-time sound intensity.

Dual microphone audio device reduces user voice interference by adjusting sensitivity to amplify distant voices.

Crossover and b-chain processors adjust frequency response and time alignment to correct asymmetries between unmatched transaural loudspeakers.

A tinnitus treatment device generates sound stimuli based on residual inhibition curves to suppress symptoms.

A speaker device uses dynamic equalizer correction to adjust frequency characteristics based on installation orientation.

A speech processing system selects pre-trained models based on speaker location and microphone array orientation to handle dynamic acoustic variations.

A sound-reproducing apparatus calculates test environment spatial parameters from received audio data to modify input signals for accurate playback.

Synchronizing and mixing internal and external audio signals prevents silence caused by unexpected microphone disconnections.

An audio processing device generates auxiliary mixtures from microphone signals to synthesize output with desired source gain factors.

An assistive listening device merges bone and air conduction sound waves to enhance perceived sound intensity for users.

Host system dynamically lowers speaker volume when hard disk performance drops, preventing vibration-induced data errors without extra hardware.

A vehicle communication system uses wind and light signals to alert pedestrians without opening windows.

Orientation sensors determine on-head status by analyzing common orientation changes, resolving detection reliability issues across varying ear shapes.

Information processing device detects users outside camera angle and generates targeted notifications for remote participants.

Microphone arrays and neural networks separate overlapping speech to improve transcription accuracy in complex environments.

A headphone system modifies audio output using sensor data to detect environmental conditions.

Dynamic beamformer switching resolves nonlinear echo contradictions by adapting filters based on real-time acoustic conditions.

Segmenting spatial features reduces misjudgment probability in noisy environments while maintaining angle estimation accuracy.

A two-wire microphone uses an ultrasonic generator for remote device authentication via shared signal and ground pins.

Constrained beamformers adapt parameters selectively based on difference measures to resolve convergence speed and speech distortion trade-offs.

A charging case touchpad detects user gestures to control wireless earbuds.

Audio analysis system determines room transfer functions using microphone arrays to estimate acoustic properties for virtual reality environments.

A voice activated assistant device uses microphone triangulation to identify audio source locations relative to the hardware.

A time-frequency conversion unit calculates spectral changes between audio frames to identify babble noise.

Dynamic Sound Pressure Limiting detects acoustic reflexes to adjust volume levels, preventing noise-induced hearing damage.

A playback device processor automatically switches audio sources between media devices without user intervention.

Audio modification system calculates user head dimensions using signal strength and latency between earpieces for personalized 3D audio rendering.

Rotatable microphone holders switch between inward and outward positions, reducing phase difference without complex sliding mechanisms.

A microphone array adjusts its beamforming direction based on detected person positions in the vehicle interior.

Jack detection circuit manages voltage states on microphone electrodes to prevent audio noise during connector insertion.

A microphone array uses a four-channel serial peripheral interface to synchronize clock signals and reconstruct interleaved data from multiple digital microphones.

A prediction system derives psychoacoustic metrics from audio signals to assess spatial quality attributes like envelopment and source location.

A sound field control apparatus calculates filter coefficients using spatial transfer characteristics to generate specific sound pressures in targeted areas.

A sound masking device mounts to existing hotel televisions using a dedicated bracket that conceals the unit and utilizes the TV power supply.

Directional microphones identify external sound sources and trigger haptic cues on the steering wheel to alert drivers when auditory signals are ineffective.