Electrophonic Musical Instruments TRIZ Cases

Progressive switching between cascaded ARMA filter cells smooths IIR filter changes, cutting latency and audible clicks in digital audio.

Harmonic bass enhancement is combined with compression and control to boost perceived bass while limiting distortion and speaker excursion.

Audio output is filtered with sound-pressure-based equal-loudness curves to keep tonal balance consistent as volume changes.

An FIR filter compensates mute-altered frequency response so wind players hear a more natural instrument sound with less harsh noise.

Subband envelope shaping preserves timbre and spectral coherence when polyphonic audio is transposed or adapted to a new key.

Fade-in and fade-out on a converted multi-bit signal suppress spike noise, making jog-dial search in one-bit audio more reliable.

Direct drive-signal and microphone capture improves impulse response measurement of amplifier-loudspeaker sound coloration and resonances.

By linking overtone bands to their fundamentals, this case enables pitch and key changes in polyphonic audio while preserving timbre and spectral coherence.

Shared analysis and synthesis filter banks support multiple transposition orders, reducing HFR audio coding complexity while preserving quality.

Discrete switch positions blend filtered tone signals to expand selectable timbres while improving repeatability over slider-based selectors.

Sampling-set differences extract modulated-signal envelopes accurately with low circuit complexity and strong resistance to DC offset.

Residual waveform coding uses common effective bit lengths to cut hardware load while preserving sound quality during abrupt waveform changes.

Sampled-signal multiplication and addition detect modulating-signal envelopes accurately while avoiding complex analog filter circuits.

Peer audio nodes combine local I/O and mixing to cut network latency, scale linearly, and avoid central mixer failure.

Soft clipping, sag control, and higher output impedance give a solid-state guitar amplifier tube-like warmth and dynamic response.

A differential CMOS distortion circuit splits audio into balanced paths to cancel asymmetry and beat notes while emulating overdriven tube-amp sound.

Dispersive waveguides replace costly impulse-response convolution to emulate spring reverb with accurate modes, disturbances, and smooth parameter changes.

Selective gain on positive and negative waveform portions restores lost highs and low-end vitality in compressed audio while limiting distortion.

Touchscreen level meters combine signal level, fader position, and channel linkage cues to assign more channels without widening the mixer panel.

A modified feedback-delay network uses Hadamard and FFT-based matrix transforms to create de-correlated multi-channel reverb with lower computation.

Non-linear feedback in a differential microphone amplifier keeps even harmonic distortion controllable across wide input signal levels.

Electroactive key actuation recreates snapover feedback by adjusting movement to switch closure and typing speed on non-rubber-dome keyboards.

Six independent amp channels and speakers separate each guitar string, preserving natural tone while producing clearer orchestral chord sounds.

Integer coupled-form bass boost filtering keeps low-frequency response stable and accurate on processors without floating-point math.

Physical controls map through screen objects to channel strips, preserving mixer assignments across input device changes with visual feedback.

Measured reverberation times guide automatic absorption-coefficient tuning in a 3D venue model for more realistic sound system simulation.

Low-pass channel cross-mixing and all-pass reverberation create headphone surround with lower processing load and adjustable spatial effects.

Precomputed filter coefficients vary with frequency and signal strength to reproduce acoustic piano harmonic decay with low IIR complexity.

A dual-output microphone amplifier switches between V2V and low-noise I2V modes, cutting resistor noise while preserving EMC protection.

Interpolated write-pointer modulation enables fractional delay lengths with lower access latency and fewer audio artifacts in DSP delay lines.

A separate field coil varies speaker sound pressure over a 25 dB range while preserving amplifier overtones and spectral content at lower volume.

Shared analysis and synthesis filter banks map modified subbands into high bands, reducing HFR complexity without extra transposers.

Block-based impulse response estimation separates direct and reverberant audio, enabling natural reverb adjustment without measuring the room.

Harmonic transposition turns inaudible bass into audible harmonics, boosting perceived bass on small loudspeakers with limited low-frequency output.

A vocoder and staged diffusion delays turn one voice into realistic, customizable crowd chants without costly live recordings.

Adjusting a separate field coil voltage lowers sound pressure while preserving the natural overtones of an overdriven amplifier.

Tracks waveform sample position so an AWG can resume stored signals at the right offset and keep phase coherence during waveform switching.

A cathode follower output stage and series attenuator cut volume while preserving overdriven tone, impedance behavior, and speaker feel.

Selective low-frequency compression adds harmonics for stronger bass on small speakers while limiting waveform distortion and processing load.

Recorded sample audio guides linear filtering and nonlinear processing to recreate studio sound engineering effects across instruments.

Upward then downward pitch transposition lets a smaller acoustic reverb element reproduce the response of much larger reverberation hardware.

Common analysis and synthesis filter banks support multiple transposition orders for high frequency reconstruction with lower complexity.

Filterbank-based envelope shaping modifies polyphonic pitch and key while preserving timbre, spectral coherence, and natural sound quality.

Transparent pad openings reveal tone icons beneath each hitting surface, so players can identify assigned sounds without test-hitting.

Bandpass harmonic locking links fundamentals and overtones so pitch, key, and mode changes preserve timbre and spectral coherence.

Frequency aperture arrays turn streamed or unpitched audio into polyphonic output with recognizable pitch and timbre in real time.

A modulation signal keeps class-D amplifier pulses wide at near-zero input, reducing EMI below 30 MHz without muting audio.

Feedback-delay networks and discrete echoes simulate complex multi-resonator reverb with lower computational load for real-time audio.

Visual spectral editing lets users shape each track’s target frequency balance while the mixer automatically adjusts EQ settings.

By shifting audio to a higher clock frequency during processing, this case boosts loudness while preserving dynamics and reducing distortion.