Signal Processing TRIZ Cases

An infrasonic cooling signal replaces heat-heavy low frequencies to cool an overheated speaker quickly without major audible quality loss.

Multiple acoustic-electric transducers with different frequency responses split audio into sub-bands faster and with less noise than DSP filtering.

Acoustic frequency transitions let wireless audio devices measure distance accurately using existing speakers and microphones, avoiding extra ranging hardware.

A ring-shaped neck-worn camera places four microphones around the neck to capture 360-degree sound without added bulk or visible mic modules.

Blending source and destination space impulse responses enables smooth virtual environment audio transitions without comb filtering.

Precomputed filters matched to speaker characteristics and volume levels prevent distortion while preserving sound quality across different speakers.

Motion data updates virtual speaker position and angle in real time, improving spatial audio follow-through and reducing fixed head-in effects.

Mixture modeling of sound-source directivity cuts transmission data and delay while preserving 3D audio directional quality.

A control audio signal marks interfering directions so the microphone array can exclude them, reducing false focus and improving audio quality.

Real-time listener tracking with video and infrared sensors lets home theater speakers recalibrate sound fields for synchronized playback.

Ambient sound monitoring matches the loud Bluetooth speaker and triggers volume reduction to keep noise within decibel limits.

Position-dependent HRTF compensation preserves spatial accuracy in binaural hearing pre-processing while supporting direction-sensitive noise reduction.

Independent control of sound-generating units shapes zone-specific sound fields, giving multiple users more consistent audio reception.

Scale-dependent nonlinear filterbanks create phase-coherent harmonics so limited speakers can mimic out-of-range frequencies with lower power.

A single-chip neural pipeline applies noise reduction before WDRC to cut latency and power use while preserving audio quality in ear-worn devices.

Delay processing keeps bone- and air-conducted sound within 100 microseconds to prevent stereo deviation and improve mid-band listening.

Frequency-band ratio detection adjusts hearing-device output only where ambient and emitted sound overlap, reducing comb filter artifacts.

Neural-network earphone detection combines voice relatedness, intelligibility, context, and direction to avoid false volume drops.

Adaptive filtering estimates acoustic feedback, then smooths frequency peaks to suppress multi-frequency howling with low sound quality loss.

Weighted sound and image scoring tags video segments after specific sounds, making desired subject states easier to retrieve.

Multiple microphone and beamformed audio inputs feed one neural network to isolate a target speaker and reduce noise across frequencies.

Separate DSP modules clean both transmitted and received audio, solving the gap left by uplink-only noise reduction.

Built-in microphones capture component noise signatures to detect electrical faults automatically without added diagnostic hardware.

A head-worn microphone array separates audio capture from camera noise and uses directivity control to record clearer sound during video shooting.

Sensor-based folding detection adjusts microphone audio during video calls to preserve the speaker's voice while suppressing ambient noise.

A high-pass crossover above 6 kHz splits dual-speaker earphone signals to curb diaphragm distortion and improve sound quality.

Scalar limit values synchronize impulse-noise attenuation across both hearing aids to preserve interaural level differences and spatial hearing.

Internal microphone feedback and transfer functions help earphones reproduce external sound naturally while compensating echo and sound leakage.

Listener tracking with cameras, infrared, and depth sensors enables real-time sound field calibration as the user moves.

Targets TWS earphone feedback by classifying howling types and combining band gain reduction with adaptive and nonlinear filtering.

Shared microphone capture and AI audio generation let connected speakers deliver more voice services with less hardware cost and interference.

Dynamic thermal control predicts speaker heating and adjusts bass and limiter settings to preserve loudness, timbre, and power use.

Gradually raising ambient noise suppression when AEC starts helps microphone processing reduce echo artefacts and abrupt audio level shifts.

Polynomial curve fitting turns a few EQ inputs into richer parametric controls, simplifying audio customization while keeping volume levels consistent.

Randomly split and crossfade real acoustic impulse responses to create diverse training data for more robust audio dereverberation.

Corrected sound measurements across nearby electronic apparatuses improve wakeup command targeting despite installation and hardware variation.

Dynamic mixing raises dialogue against music and effects at low volume or noisy settings to improve speech clarity without manual rebalancing.

Uses vehicle amplifiers and exterior loudspeakers for high-power outdoor audio, then disables playback automatically when the vehicle moves.

A telematics-controlled switch reroutes an existing vehicle speaker for emergency calls, preserving two-way audio while reducing extra hardware.

Uses vehicle loudspeakers and an amplifier for stronger outdoor sound, then disables playback automatically when motion is detected.

Audio, IMU, and user context are ranked in real time to restore spatial awareness during ANC and surface critical sound events.

A linear AEC plus DNN residual canceller improves echo suppression accuracy, convergence speed, and computing efficiency in audio calls.

Magnetic field wave sensing replaces destructive sampling and subjective appraisal to rapidly assess artifact age and authenticity.

Sensors and head-position tracking separate in-vehicle audio streams, reducing interference, distraction, and unsafe volume changes.

Multiple microphone signals suppress dominant noise sources while complementary audio restores the scene for a clearer spatial image.

Spatial weighting of spherical harmonic sound-field maps improves source localization and separation in reverberant microphone-array recordings.

Switching a wireless headset between bi-directional and uni-directional audio links preserves media playback clarity during calls.

Loudness metadata and ML-based peak estimation set adaptive gain for speaker playback, keeping volume consistent while avoiding clipping and distortion.

Sensors track user proximity and background noise so playback volume and wake-word thresholds adjust to improve clarity and cut false positives.

Pre-synthesis level matching balances multiple sound sources by frequency peak, preventing one voice from overpowering others in distributed content.