Signal Processing TRIZ Cases

Real-time sound adjustment compensates for device variation, keeping amplified or attenuated frequencies within each user's hearing range.

A hearing device detects impulses in selected frequency bands to reduce discomfort without distorting temporal speech characteristics.

A level detector before synthesis and limiter after it curb loud-output distortion and voltage-dip malfunctions.

Virtual acoustic models reduce repeated room testing while preserving consistent audio evaluation.

Selective neural network loading improves hearing clarity while reducing power and memory use.

Context-aware device selection synchronizes voice responses across networked playback zones for coordinated control.

A filter bank, upstream level detector, and downstream limiter stabilize hearing device output during loud transient noise.

The speaker measures impedance at different frequencies to detect resonance-space changes and adjust gain for more stable sound output.

A sensor compares amplified and initial audio signals to detect playback issues, tampering, and volume needs.

A mixed-rendering audio system combines far-field output with non-occluding near-field speakers for personalized, intelligible speech.

This case uses test signals and microphone responses to adapt DSP settings as room layouts and device placement change.

Frequency-band sub-signals drive multiple transducers to adapt spherical sound dispersion across changing listening environments.

This audio processing case uses images to detect room type, then adjusts gain and noise suppression without manual setup.

This case coordinates networked playback devices by detecting audio sources and changing playback characteristics to limit interference.

Volume-dependent Linkwitz transforms extend microspeaker bass while compression limits distortion.

This case uses slotted calibration audio, delay measurement, and digital filters to align drivers across listening environments.

Scene recognition automatically adapts smart-device audio settings across environments.

Ambient-noise switching balances receiver and screen audio, reducing quiet-call leakage while preserving clear communication in noise.

This acoustic system filters and synthesizes sensor signals to minimize speaker components while preserving target sound capture.

Weighted filter interpolation tracks six-DOF head movement for real-time crosstalk cancellation.

Mapped transfer functions adapt speaker filters to user position and orientation, preserving left-right audio separation during movement.

This audio playback case separates frequency bands and applies tailored delay and gain filters for clearer crosstalk cancellation.

This case uses ipsilateral and contralateral spectral levels and symmetry to detect own voice without specialized components.

A processor uses environmental microphones and context-specific models to adjust speaker volume without manual intervention.

A neural model turns noise into spatially encoded audio, reducing reliance on complex physical models for varied XR sound.

The audio system maps user position and orientation to adaptive filters that reduce spectral distortion during head movement.

Integrated sensors detect hearing aid shifts and adjust sound processing.

Connection-aware audio switching routes sound privately and mutes output on command without interrupting displayed content.

Multiple sound sensors and frequency-band processing remove speaker-derived components while preserving useful ambient sound.

The case separates multiple sound sources, adjusts angle and distance for each listener position, and mixes realistic stereophonic audio.

This case separates audio bands, applying midband cancellation and delay filtering for accurate left-right sound delivery.

Precomputed transfer functions and nearest-point weighting reduce computation for real-time crosstalk cancellation during head movement.

This case uses coil electrical data and a thermal model to predict panel temperature and adjust audio output before overheating.