Speech Analysis TRIZ Cases

A speaker classification system extracts voiced speech frames and computes mel-frequency cepstral coefficients for feature extraction.

A signal processing device uses semi-supervised nonnegative matrix factorization to separate sound sources and calculate activation levels of spectral bases.

An adjustable shelf filter and attenuation block process microphone signals to reduce uplink noise in mobile devices.

A dual-module audio processing system combines spatial cue analysis with neural network source separation to generate clean output signals.

A generative model calculates voice propagation distance to draw a dynamic virtual circle, transmitting audio only to participants within the radius.

A control unit compares incoming acoustic signals against stored sample patterns to generate adaptive display information.

A neural network classifier accesses a pre-encoded database to generate standard-compatible bitstreams, reducing runtime processor usage and memory consumption.

A perceptual weighter and quantizer apply a random matrix to an input signal spectrum.

A vehicle audio filter processes sensor signals via short-time Fourier transform to enhance target speech clarity.

Integrating differentiable digital signal processors into machine learning models enables gradient-based training, reducing computational energy costs.

A hearing aid controller uses a trained machine learning model to dynamically adjust sound processing settings based on detected environmental properties.

A noise reduction apparatus detects speech segments and voice incoming direction to process audio signals from multiple microphones.

Wearable earpieces map external auditory canal structures to generate unique identifiers, reducing unauthorized access risks in wireless transactions.

Weights frequency bands by disturbance presence to preserve user speech elements and improve discrimination accuracy.

A contact center system captures call pickup timing to distinguish live agents from automated voice responses.

A processor extracts input feature data and calculates matching scores using common component vectors to identify enrolled users.

Client-server speech recognition systems encrypt medical dictation data before transmission and manage decryption keys locally to protect sensitive information.

Mobile sensors capture occupancy and noise data during user check-ins, resolving the trade-off between search result accuracy and data staleness.

Audio transmission apparatus generates playback environment information and encodes three-dimensional audio signals for efficient delivery.

A deep adaptive acoustic echo cancellation system integrates a neural network with linear filtering to generate nonlinear reference signals.

Alternates predictive and transform encoding to resolve the trade-off between coding quality and algorithmic delay.

Extracts optimal coding parameters to minimize spectrum quantization error and enhance synthesized speech quality.

Weighted pitch prediction estimates lag values using reliability metrics to reconstruct speech signals during frame losses.

Segmenting audio links allows independent latency tuning, resolving the conflict between wireless robustness and prompt user interaction.

A packet loss concealment method combines substitute signal differences with dequantized prediction errors to generate a combined transition signal for ADPCM decoders.

A decoder classifier distinguishes speech from non-speech audio signals to enable selective noise suppression.

Circuitry separates noise components from monitoring sound data using independent component analysis to generate cancellation signals.

Audio encoding decomposes signals into subbands with differential feature dimensionality to balance compression efficiency and fidelity.

Encoder-generated side information guides transcoders to skip intensive encoding processes, reducing computational complexity while preserving audio quality.

An audio encoding method generates a predicted current frame signal by synthesizing phase and gain differences from previous frames to output a reconstructed residual.

Adjacent pulse pairs in a dual-pulse excitation model reduce bit rates and complexity while maintaining perceptual quality.

A device adapts psychoacoustic audio decoding to operating conditions by selectively processing scene-based audio data subsets.

Segmenting transient and non-transient signals resolves temporal smearing artifacts while maintaining high perceptual quality at low bitrates.

A speech detector isolates vocal segments in upmixed audio signals, suppressing them in back channels to prevent incorrect source localization.

Microphone sensor data triggers aerosol generation only when manual actuator input confirms user intent, preventing accidental activation.

Two sequential sweeps determine an optimal pass-band and apply filtering to suppress ambient noise and reduce pre-response contamination.

A hearing aid signal processor transposes frequency bands to render inaudible sounds audible while preserving harmonic coherence.

Local speaker recognition processes voice biometric prints on mobile devices to eliminate bandwidth consumption from remote server transmission.

A talker identification unit integrates voice quality models with auxiliary data for precise user recognition.

Audio interface enables voice-based ATM transactions, verifying selections to prevent privacy leakage.

A local device executes pre-configured stimulus-based actions independently of network connectivity to maintain scheduled notifications.

A method pauses audio playback when signal power falls below a threshold.

A hybrid expansive frequency compression method remaps speech frequencies to enhance auditory perception.

A transmission apparatus detects trigger conditions in voice or text sessions to automatically acquire and send emoticon images.

Automated reanimation adjusts video frame counts to match dubbed audio duration and synchronizes character mouth movements with vocal sounds.

Scrambled voice fragments protect privacy during external analysis by preventing reconstruction of original speech.

A parametric audio processing method synthesizes multi-channel output signals using base channels derived from input signals and a coherence measure.

A streaming matching system ranks audio probe samples against reference data using continuous confidence scoring to identify relevant matches.

A speech reproduction device generates adaptive masking sound to render speech unintelligible in a masked zone.