Speech Analysis TRIZ Cases

A signal processing apparatus converts input signals into phase and amplitude components in the frequency domain to calculate phase linearity.

Correlating microphone and accelerometer signals distinguishes enrolled users from impostors, resolving text-independent authentication reliability issues.

Fourier transformation on framed audio segments detects compression distortions to distinguish true lossless files from false lossless formats.

Virtual channel generation separates coherent signals from miniaturized microphone arrays, resolving positioning errors caused by device size reduction.

Envelope shaping preserves temporal envelopes during binaural audio synthesis, reducing pre-echo artifacts and washed-out transients.

A termination burst carries a data synchronization pattern and slot type field to signal call end within a multi-burst superframe.

A predictive audio codec divides signal blocks using a mid-block start state to encode parts in opposite directions.

Comparing discrete sound class labels from dual microphones reduces false positives in audio recognition systems without complex signal subtraction.

A multi-modal learning platform integrates speech analysis and interactive modules to enhance language fluency.

Adaptive pitch period encoding switches resolution and mode according to signal stationarity, reducing information complexity while maintaining coding quality.

Segmented multi-task training with ArcFace loss optimizes feature vectors, resolving accuracy drops when handling varying speaker counts.

A layered audio coding format decomposes signals into rotated components and gain profiles to enable efficient encoding of monophonic and sound field representations.

A universal decoder structure processes excitation signals from both Code Excited Linear Prediction and Vector Quantization Noise Feedback Coding encoders.

A vehicle system detects driver emotions via voice and image data to deliver targeted pacification content.

A signal separator processes microphone inputs to isolate useful audio content from mixed sources.

A composite audio signal processing method combines source separation with coherence-based envelope filtering to enhance speech quality.

Wireless audio streaming between handheld game devices uses real-time format conversion to deliver continuous sound playback.

Encodes location and energy of tone components to resolve bandwidth quality trade-offs.

A codec selects coding modes based on spectral peak distance and noise floor ratios.

A geometry-aware landmark generation network predicts emotion-driven facial landmarks from speech audio and independent emotion inputs.

Auto-regenerative neural network reconstructs high-quality wide-band speech from narrow-band parametric parameters, reducing data transmission volume.

Space-depth transformation with ASPP reduces model size while preserving receptive fields for efficient music source separation.

A vehicle sound processor tunes and mixes audio signals across distinct frequency bands to create a calming acoustic environment.

A neural network model identifies additional samples to upscale digital signals in wireless devices.

A document identification system extracts image frames and audio signals from video clips to verify physical ownership.

An audio system adjusts parametric noise levels across frequency bands to enhance speech intelligibility.

An arithmetic decoder adjusts context states using previously decoded spectral values to optimize mapping rules.

Neural networks separate speech from multiple sources using brain signals, reducing device complexity and latency.

A data generating apparatus converts real-world audio into short-time frequency spectra to train deep neural networks for noise elimination.

A feedback delay network applies frequency-dependent late reverberation to multi-channel audio for binaural virtualization.

A system detects audio nuisances using spectral features and multi-frame tracking to notify users of unwanted sounds.

A vehicle controller synthesizes driving sounds by analyzing facial images and biometric signals to match driver preferences.

A computer system detects registered users and privacy threats using sensors to adjust its operating privacy level.

A voice encoding device selectively quantizes frequency bands to improve coding efficiency.

A speech inpainting method synthesizes missing audio segments from text transcripts and applies voice conversion to match the original speaker.

A Bark-scale spectral partition method rotates phases at identified peaks to maintain coherence in critical frequency bands.

Latent descriptor vectors transform multiple-valued output problems into single-valued outputs by adjusting model parameters based on generated vectors.

Partly masked in-domain data elements optimize encoder parameters, resolving domain mismatch between simulated signals and real-world audio inputs.

Neural audio encoder separates pitch data from symbolic representations to compress speech at low bitrates while maintaining high quality.

A speech decoder reconstructs erased frames using adaptive codebook gain values derived from preceding signal data.

Segmenting acoustic signals into energy sequences allows parallel differentiators to classify speech locally, resolving latency and power constraints.

Combining adaptive and fixed filters reduces broad frequency noise without increasing device complexity.

Configurable upsampling separates ACELP and SBR domains to resolve temporal resolution limits at mid-bitrate ranges.

A noise suppression system corrects estimated a priori signal-to-noise ratios to calculate dynamic suppression coefficients for input signals.

A data positioning apparatus detects discrete audio timings using segmented display regions for precise selection.

Encoder embeds loudness metadata into bitstreams to eliminate pumping artifacts from real-time leveler divergence.

Blind Sound Source Separation algorithms remove background noise from audio input signals generated by proximate agents.

A speech signal processing circuit calculates spectral balance ratio features from time-frequency domain components to determine perceived audio quality scores.

A processing device attenuates repeated packets to conceal real-time data loss.