Speech Analysis TRIZ Cases

A quantization unit selects from multiple signal-to-noise ratio quantizers to optimize voice encoding.

Variable frame durations across base and enhancement levels resolve suboptimal compression ratios by allowing dynamic bit stream formatting.

Segmenting audio decoding into independent priority paths maintains high sound quality without increasing conversation delay time.

A transmission device divides signals into data elements and drops inactive frames during stable periods to maintain signal integrity.

A multi-channel audio encoding method calculates spatial masking thresholds to determine per-channel bit allocation.

Dynamic rotation angles adapt to phase offsets, resolving compression quality trade-offs in stereo coding.

Automatic pairing based on angular direction estimates admits only relevant audio signals, reducing network congestion and power consumption.

Reduces high bit rates for mobile streaming by encoding truncated HOA representations with frequency-specific directional predictions.

A failure detector monitors sound intensity level differences between microphones to identify degraded units.

A speaker recognition system pre-filters speech signals using spectral analysis to discard noise and non-speech frames before feature extraction.

A system measuring electroencephalogram signals to determine speech sound discriminability without manual user input.

A signal analyzer selects optimal analysis windows from multiple options to improve perceptual audio quality in bandwidth extension encoders.

Host analyzes sound signals to generate on-screen display patterns, resolving the inability to interact with display settings without physical buttons.

Continuous listening device analyzes audio conversations to automatically adjust media asset priority, eliminating manual configuration effort.

Interpolated primitive matrices blend discrete spatial configurations to eliminate zipper noise artifacts during rapid object movement.

A signature matcher compares monitored media signatures to reference streams using noise cancellation models.

A transmission terminal negotiates sampling rates and bit-rates with a reception terminal to compress voice signals dynamically.

Anti-sparseness processing fills spectral holes in low-frequency bands, reducing metallic noise during high-frequency extension.

A hearing device applies adaptive frequency transposition to shift signal components based on momentary input characteristics.

Frequency-domain sub-band weighting updates separation matrices to isolate audio sources, reducing sensitivity to microphone position errors.

An adaptive high-pass filter reduces coding noise in decoded audio signals by targeting frequencies below the fundamental pitch.

A high-band encoding method generates sub-band-specific bit allocation information based on a low-band envelope to refine spectral data.

A media processing system identifies individuals using speaker and image recognition to automatically tag pictures and videos.

Inverse weighting and segmented search reduce computational complexity in closed-loop CELP processing while maintaining high coding accuracy.

Preprocessing apparatus detects frame characteristics to selectively remove long term correlation, reducing coding complexity for silence and unvoiced signals.

Decomposing lung sound signals isolates crackle components from respiration noise using empirical mode decomposition.

A wearable electronic device uses a processor to generate beamformed signals and configure filters with passbands at integer multiples of fundamental frequency values.

Auditory manager dynamically adjusts sound levels using machine learning to mitigate interference from distance, objects, and humidity.

Frequency-domain spatial audio coding uses time-frequency direction vectors to describe the audio scene independently of channel configurations.

A computer-implemented method assesses acoustic performance by analyzing captured audio signals to guide user adjustments.

A generator and discriminator network transforms noisy feature vectors into denoised representations for improved classification.

A hearing aid sampling rate converter adjusts clock rates via PID control to resolve synchronization drift caused by lost data packets and clock jitter.

Master radio sets segment communication roles to enable seamless voice command transfer across tactical teams.

Automated system selects and arranges player activity data into composite streams without manual intervention.

A linear predictive analysis apparatus adjusts autocorrelation coefficients based on fundamental frequency and pitch gain to enhance spectral envelope representation.

Adjusts spectral data to a reference shape, resolving narrowband signal measurement errors and improving accuracy.

A speech decoder shapes high-band temporal envelopes using linear prediction coefficients to improve decoded signal quality.

Landmark-based convolution processes sparse sound data to reduce computational resource consumption while maintaining high pattern identification accuracy.

A signal processing apparatus shifts decoded audio phase using inter-channel phase difference flags to reconstruct spatial sound fields.

A channel selection mask directs an audio decoder to process only necessary channels from a bitstream.

Embedding digital tags in audio streams enables automatic detection of reception status, resolving uncertainty about whether participants are being heard.

Automated acoustic analysis of breathy phonation and prosodic features distinguishes suicidal from non-suicidal subjects with high accuracy.

A cloud-based system verifies user identity by comparing reflected audio signals against stored acoustic tokens during video sessions.

A deformatter unit separates scene description data from object audio signals for individual processing.

Automated post-conversation evaluation analyzes sentiment and tone to resolve agent engagement trade-offs.

Apple-peeling codebook construction reduces computational load, enabling low algorithmic delay in wideband audio encoding.

A coding mode determination apparatus selects audio signal modes using stored historical frame characteristics.

An audio noise reduction selection computing module configures rules to apply specific models.