Signal Processing TRIZ Cases

A mobile cluster-based audio adjusting system uses wearable sensors to autonomously tune sound outputs.

A user device measures audio chain latency by capturing speaker output with a microphone and sending adjustment data to components.

Processor generates harmonic signals from high frequency bands and overlaps them with original audio to adjust levels.

Adaptive audio classification derives acoustic features to differentiate speech from noise using inter-microphone level differences.

A calibration system determines an inverse transformation of room acoustics to restore distorted speech signals, enabling reliable hands-free voice interaction.

Coherence processing between microphone signals estimates environmental noise levels, reducing multichannel adaptive filtering complexity.

Audio signal processing stage separates input signals into frequency bands and compresses them to prevent loudspeaker overdrive.

A wireless device detects user and loudspeaker positions using internal microphones to steer audio signals for surround sound.

A signal processing device aligns speaker amplitude and phase characteristics to synchronize sound arrival times across multiple listening points.

Camera and auxiliary device process images to track speaker location, improving speech intelligibility while reducing battery consumption.

A hearing aid method divides audio signals into frequency bands to apply targeted distortion.

A vehicle microphone cluster uses signals from user electronic devices to create communication zones.

An adaptive beamformer adjusts real-valued weights to combine microphone signals based on individual signal quality.

Comparing main and auxiliary microphone volumes allows dynamic adjustment of acoustic filtering, ensuring voice signals exceed preset volume thresholds.

Processing circuitry detects external sound amplitude differences across multiple microphones to generate user notifications.

Segmented directional processing controls peak levels from unwanted sources while preserving intelligibility of desired signals in complex environments.

A sound quality device measures voltage and current to estimate impedance and adjust resistive values in the audio path.

Processor analyzes audio spectral content to detect user facing direction, automatically unmuting the microphone when intended speech is detected.

Processor applies a time-dependent gain decay envelope to reduce audio content levels, preventing hearing damage while preserving perceived loudness.

A hearing device directional system equalizes beamformed signal phase and amplitude to preserve directional cues.

Eigenfilter theory minimizes mean square error to flatten frequency responses while preventing speaker non-linearity.

Opposing speaker diaphragm vibrations cancel shell mechanical noise, preserving bone conduction microphone voice input quality.

A sound data processing system transmits positional and directional data to determine source location.

A dynamic token algorithm manages primary and secondary speaker roles in a Bluetooth network to enable true wireless stereo audio communication.