Signal Processing TRIZ Cases

By comparing dry and wet signal envelopes to playback limits, this limiter preserves bass impact while avoiding choppy loudness shifts.

Pre-scaling and dynamic range remapping adapt audio signals to user volume and speaker limits, preserving bass, clarity, and low distortion.

A unified control approach lets playback devices switch between audio sources and keep volume settings consistent across connected speakers.

Dual-stage attenuation controls fast acoustic shocks in hearing devices, limiting discomfort and distortion while keeping the event audible.

Dynamic noise dose tracking uses earpiece calibration and audio content to adjust output levels and reduce hearing loss risk.

Switching between voltage and current source drive cuts electromagnetically coupled noise at low volume while preserving hearable output power.

Audio components below hearing thresholds are removed after micro-speaker response filtering to cut wasted power and prevent voice coil overheating.

Dual-stage attenuation around an output limiter controls acoustic shocks quickly while preserving natural sound awareness and comfort.

Frequency-response feedback verifies headphone or earphone fit before self-measurement, improving ear canal transfer accuracy.

Periodic model-transistor tracking cuts quiescent overload-protection power in Class D amplifiers while preserving accurate fault detection.

Audio equalization lets an LRA handle low-frequency speech while limiting resonant peak distortion and helping mobile devices meet 3GPP response masks.

Dynamic low-frequency adjustment based on detected volume keeps small speakers bass-rich across source changes and low listening levels.

Proximity sensing switches the main and secondary microphones automatically, improving voice clarity and noise cancellation with manageable control complexity.

Limits analogue gain to the lower safe level and shifts gain digitally to preserve dynamic range without raising the noise floor.

Ambient sound sensing identifies dominant noise or speech so a terminal can automatically lower or adapt playback volume for clearer, more comfortable guidance.

Separate digital audio paths with adaptive analogue gain preserve music dynamic range while limiting noise and THD when system sounds are added.

Measures room frequency response with a smartphone microphone by adapting test signal volume to background noise and generating playback correction.

Built-in microphones detect phase differences and room acoustics to auto-correct speaker frequency response without manual calibration.

Dynamic amplitude-based volume adjustment adapts playback to audio level, user context, and noise while reducing terminal power use.

By combining external noise with user position and motion, this case uses time windows to avoid abrupt earphone volume changes.

Precomputed lookup tables and indexing speed audio filter gain retrieval, cutting runtime load in VR and AR magnitude response control.

Direct and reverberant sound are mixed by zoom level and object direction so audio matches the zoomed image and reduces viewing discomfort.

Equal-loudness weighting lets one monitor pair mimic different speaker grades, helping mixing engineers judge playback across systems.

Dynamic audio processing uses amplifier condition and load impedance to cut power use, protect against heat, and preserve sound quality.

Multiple DSP algorithms and controllable amplifier channels let one sound box adapt to room layouts and changing user sound preferences.

A codec compares parallel audio inputs and applies internal gain to avoid volume-control delay while preserving signal-to-noise ratio.

Environmental noise sensing lets audio volume rise gradually to keep notifications audible without causing unnecessary disturbance in quiet settings.

Source-aware volume control lets PTT radios raise or suppress voices by communication source, reducing missed messages and unwanted audio.

By shifting audio into a speaker's peak output band, this case boosts small-device volume and speech clarity in noisy settings.

Variable FIR filtering keeps loudness equalization in the time domain, avoiding FFT-driven complexity and latency in audio processing.

Converting sound pressure by frequency band into loudness levels enables audio adjustment that better matches hearing perception and limits harmful volume.

By comparing drive and voice-coil current signals, one speaker can play audio and detect ambient noise without separate microphones.

Masking-curve-based multi-band compression adapts audio to a user's hearing profile, improving dynamic range and perceived sound quality.

An interference-resistant branch uses impedance matching to cancel analog switch coupling and protect digital signals on a shared interface.

Hearing profiles drive multiband compression, gain, and dynamic range tuning to improve speech intelligibility and sound quality.

A single control interface coordinates source switching and preserves volume settings across multiple audio playback sources.

Built-in microphones detect reflections and switch EQ profiles automatically when a playback speaker moves between indoor and outdoor spaces.

Maps volume levels to dynamic range so audio processing preserves audible detail and adapts sound output to user hearing characteristics.

Gradual gain ramping before and after signal parameter switching keeps volume changes within limits and avoids audible user discomfort.

Selectable amplifier gain and digital bit shifting help a microphone handle varied sound scenarios while preserving dynamic range and signal scaling.

A threshold-triggered compression block limits short audio peaks to cut distortion and protect small speakers without complex dynamic algorithms.

Spectral subband event detection in a perceptual loudness domain reduces audible artifacts while keeping dynamic gain control accurate.

Precomputed speaker filters match user audio profiles from measured frequency responses, improving personalized playback without heavy real-time processing.

Real-time noise analysis adjusts vehicle audio across spectral bands to keep sound clear without repeated driver volume changes.

Test signals and microphone feedback automatically tune multiband compressor bands and thresholds to limit loudspeaker distortion and echo leakage.

Lookup-table calibration maps one selected volume to product-specific speaker levels, keeping sound pressure balanced across mixed speakers.

Power ratios from near and distant microphones separate distance-driven level changes from natural speech dynamics for robust close-talk AGC.

Phase alignment with all-pass filtering and delay reduces destructive interference between discrete audio drivers at the listening location.

Measures time-varying levels across frequency bands and summarizes them as spectral-dynamic values for faster audio processing comparison.

A single volume interface balances music, calls, and noise-based DSP adjustments to keep multiple audio sources clear and user-controlled.