Speech Analysis TRIZ Cases

Adjusts global stereo phase parameters using smoothed inter-channel time differences across sub-bands.

A method determines inter-channel time difference by selecting local maxima from cross-correlation functions for positive and negative time-lags.

A digital audio decoding method synthesizes replacement frames by adding weighted noise to selected spectral components.

An audio encoding apparatus detects tones at frequency band boundaries and suppresses one component to prevent decoding artifacts.

Applying median filtering to raw banded gains reduces musical noise artifacts while preserving spectral selectivity.

Dynamic weight estimates preserve out-of-phase energy during down-mixing, avoiding complex phase alignment.

Encoding direction information within frequency subbands of Higher Order Ambisonics signals reduces data rates to 128 kbit/s for mobile streaming.

An apparatus adjusts temporal smoothing based on encoding metrics to optimize spatial audio rendering.

A mobile terminal application collects diverse biometric data using built-in sensors and neural networks for stress assessment.

A speech intelligibility evaluation method uses disturbance density functions to derive quality parameters from frame pairs.

A voice activity detector adjusts noise adaptation rates using spectral analysis and temporal variability to distinguish speech from background noise.

Segmenting downmix matrices into unit primitives with constrained coefficients reduces bit requirements while maintaining audio reproduction accuracy.

Frequency-domain filtering attenuates noise in spectral valleys and enhances energy near band intersections, resolving artifacts from lossy compression.

Analyzing periodic variation in speech features detects synthetic signals, preventing unauthorized access to speaker verification systems.

Subband filtering separates low-frequency components for precise delay calculation, correcting phase mismatches that distort beamforming.

A method infers experienced emotions from expressed text using likelihood probabilities.

AI synthesizes operator voice clones to deliver personalized automated responses, replacing static tone banks that lack caller context awareness.

Work-type identification bridges simple volume detection and effective noise reduction by adapting discomfort conditions.

Audio filtering modifies voice segments before transmission, neutralizing replay attacks that exploit recorded audio data.

Direct transitions between variable windows eliminate decoding delay while maintaining reconstruction quality.

A headset processor extracts speech temporal characteristics to generate real-time call metrics without full content analysis.

A monaural noise suppression system transforms acoustic signals into cochlear-domain sub-bands to track pitch sources for accurate speech reconstruction.

Alternating pitch shifts modify audio signals to suppress microphonic feedback, maintaining high volume levels without howling.

Segmenting sample generation across two time intervals reduces computational load while maintaining signal continuity during frame loss.

A coding model selection rule activates only after an analysis window accumulates sufficient audio signal sections to ensure valid buffer contents.

An extraction unit adds reproduction start times to MP4 samples within segment files.

Machine learning system extracts audio features to automatically derive new sound mix versions, resolving manual versioning bottlenecks.

A caller identification apparatus analyzes voice characteristics to distinguish between trusted contacts and fraudsters.

Separate decoding paths for spatial and augmentation audio signals reduce processing complexity while maintaining high fidelity during user movement.

Lagrangian multiplier method balances distortion and bit rate to resolve convergence issues in nested loop search.

A filter determination unit estimates power spectral density information to separate direct and ambient signal portions in audio processing.

Extracting side information representing relations between signals and objects enables remixing without unnecessary processing overhead.

Segmenting detection into low-power monitoring and high-accuracy verification stages reduces power consumption while maintaining precision.

A sequential neural network layer processing method loads parameters one by one to execute inference on constrained devices.

A noise-aware audio visual speech denoising system fuses refined audio and visual features to generate a de-noised signal.

Matching audio signals from dual microphones authenticates device proximity, reducing infrastructure complexity while preventing unauthorized access.

A UWB-based digital audio transmission unit processes signals via a radio frequency transceiving module and UWB processing module.

A processing unit detects the language with the highest word count in a display image to enable the corresponding input method.

AR glasses separate mixed audio streams using machine learning to identify and record specific sounds via user-selected icons.

A noise cancellation system injects an opposite voltage into a microphone power supply to mitigate audible crosstalk in three-wire headsets.

Periodic SID frame transmission updates noise parameters without energy threshold comparisons, maintaining connection reliability.

A model evaluates conversation quality by analyzing relationships between speech segments and turn-taking patterns.

A signal processing apparatus weights multi-channel audio signals using a center-to-total magnitude ratio to enhance voice components.

AI-driven audio enhancement system applies selective background music to customer channels.

An encoder applies pulse vector coding to a specified effective frequency range within the signal spectrum.

A wireless user equipment device captures masked audio signatures from speaker channels to determine its relative position within a spatial configuration.

An embedded browser intercepts audio streams to embed inaudible watermarks before speaker output.

A hearing aid estimates signal-to-noise ratios using recursive non-linear smoothing to refine audio processing.

A system combines quantitative data with qualitative audio and video assessments to evaluate employee performance.

A digital twin simulation models call center networks to test load-balancing algorithms in a controlled virtual environment.