Speech Analysis TRIZ Cases

A USB UART common communication system switches between signal modes using a single connector to output digital audio files.

Dynamic codec parameter adjustment reduces transmission delays and power consumption during multi-device audio connections.

Primary ambient decomposition extracts directional and ambient components from channel-based audio for higher-order Ambisonics encoding.

A conferencing system purifies audio feeds by removing unwanted components during active communication sessions.

A transient noise detection method uses pre-calculated spectral flatness values to identify and remove button-clicking artifacts from audio streams.

Neural network extrapolates signal envelope and excitation samples to generate wideband speech output from narrowband input.

Adaptive band-pass filters decompose audio signals into perceptually meaningful components for efficient parameterized representation.

Adaptive threshold detection method for voice signals reduces false detections caused by background noise interference.

Eliminates QR code scanning constraints by embedding audio watermarks in media streams, enabling passive content delivery based on device location.

A feedback recurrent autoencoder uses decoder state to generate compact codes for audio signals.

Overlapping autocorrelation sections resolve discontinuities at boundaries, stabilizing pitch tracks while maintaining computational efficiency.

Voice processing apparatus detects speaker associations through signal intensity correlation analysis.

Time-variable allpass filters adjust cutoff frequency and quality to resolve room impulse response estimation ambiguity in upmixed audio.

An audio encoder adaptively adjusts encoding accuracy based on real-time noise information to prioritize signal components less affected by background interference.

Matching extracted frequency and phase angle sequences against reference databases to detect tampering and pinpoint specific geographic locations within a city.

Fuses i-vector, x-vector, and d-vector features using linear discriminant analysis to enhance speaker verification.

Integer MDCT transform applies spectral shaping to rounding errors, reducing bit rate while maintaining lossless coding precision.

Correlating pseudorandom sequences enables reliable sample slip detection, maintaining data integrity despite vocoder modeling limitations.

Neural network upsamples narrowband voice vectors to wideband space, resolving information loss from mixed bandwidth sampling.

A hearing device extracts directional sound signals and clusters semantic representations to amplify specific conversations.

A sound output control apparatus selects portions of sound data using voice activity detection and volume moving averages.

An audio encoding method stabilizes inter-channel time difference values by reusing previous frame parameters when signal characteristics indicate environmental noise.

Collaboration applications capture facial and vocal data to enhance multi-factor authentication confidence.

A DTX decision apparatus splits input signals into sub-bands to track characteristic variations for accurate noise detection.

Stereo cameras enable eyewear diarization, separating mixed speech into individual streams for spatial audio feedback without visual obstruction.

A machine learning system extracts linguistic features from speech records to predict disease states across multiple languages.

A signal processing apparatus transforms mixed signals into phase and amplitude components to enhance output quality.

A convolutional neural network predicts speech transmission index and signal-to-noise ratio from live audio buffers.

A voice processing method reconstructs target frames using pre-extracted time and frequency domain parameters from historical data.

Magnitude-squared frequency-domain processing enables independent noise suppression without distorting the target signal phase.

Dynamic threshold adjustment resolves the reliability-fraud trade-off by enabling administrators to calibrate match sensitivity via visual interfaces.

A topological approach separates mixed audio streams using contour tree construction on smoothed weighted histograms.

An adaptive attenuation scheme adjusts signal processing based on spectral region width to maintain audio quality.

A transient detector analyzes audio signal characteristics to generate a hangover indicator for the following frame.

Adaptive block switching in a hybrid audio codec cancels aliasing during mode transitions while short windowing improves transient signal coding performance.

A personal virtual assistant negotiates future communication schedules by presenting sender options based on recipient availability.

Audio content recognition generates hash codes from discrete cosine transform coefficients to maintain accuracy in noisy and asynchronous environments.

A data correction apparatus updates reference sound signals to improve acoustic quality evaluation accuracy.

Segmenting echo cancellation into separate voiced and unvoiced neural networks resolves the contradiction between device complexity and speech quality.

Tonality-dependent filtering reduces metallic artefacts by suppressing overly strong harmonics in the highband speech signal.

A program segments audio into voice and noise frames to calculate spectral distortion amounts for precise signal quality assessment.

A deep learning system predicts blendshape weights from audio embeddings to automate avatar animation and reduce manual processing time.

Multiple beamformers isolate target sound sources by pointing to dominant spatial directions and processing outputs independently.

Audio segment comparison detects outlier peaks to assign artificial user probability for endpoint devices.

A multimodal decoder controller manages mode transitions to reconstruct audio signals from defective frames.

Visual depth maps guide audio positioning to preserve 3D space perception cues despite increased device complexity.

Dynamic radio positioning tracks the speech source to reduce background noise, improving clarity without manual adjustment.

A portable device captures audio feedback to identify broadcast media content and enable user interaction without complex hardware.

External preset metadata controls audio object levels and positions within a downmix signal.