Speech Analysis TRIZ Cases

A speech signal transmission apparatus extracts power and synchronization parameters from multichannel inputs to generate compressed extraction signals.

Multimodal embeddings combine video and audio features to characterize media items, resolving the contradiction between processing speed and detection accuracy.

Independent band unwrapping prevents noise interference, enabling accurate sound source localization.

A stereo audio encoder bypasses temporal noise shaping filters when prediction gains diverge significantly between channels.

A mobile application controlling method locks software when user authority changes.

A sound processing apparatus generates late reverberation parameters from early room impulse responses using echo histograms and spectrograms.

Reclassifies audio signals as voiced based on periodicity parameters to enable time-domain encoding.

Symmetric distribution of downmix information parameters resolves quantization errors and sound degradation in mastering signal processing.

Filterbank-channel dependent phase corrections align temporal characteristics in bandwidth extension, minimizing transient energy spread and pre-echoes.

A spam detection system compares incoming voicemail transcripts against user-generated templates to identify unwanted messages.

A multi-microphone system selects spatial processing filters to isolate speech signals from background noise.

A wearable neck-mounted device uses computer vision to recognize hand gestures and predict user actions.

Sparse audio signal processing reduces bandwidth by extracting essential inter-channel parameters from high-rate audio.

A noise suppression unit processes deteriorated audio signals using separate known and unknown noise handling sections.

A parameter interpolator generates high-resolution spatial audio data from low-resolution inputs using adaptive interpolation characteristics.

A processor estimates pitch lag using residual signal peak locations and iterative pruning to refine candidate selection.

Audio decoder extracts system metadata from MPEG-H bitstreams for transmission over HDMI interfaces.

A controller outputs situation-specific voice command guides to assist novice users with interactive display control.

Adaptive segmentation of time-domain aliased frames suppresses pre-echo artifacts while maintaining coding gain and reducing processing delay.

A speaker recognition system segments spoken phrases to extract personal identifiers and common components for simultaneous identity verification.

Dynamic FFT interval adjustment reduces analysis time for long sound data while maintaining measurement precision.

A multi-channel encoder generates down-mix output signals and complementary parametric data.

An information processing apparatus extracts maximum sound intensity values at coarse temporal resolution to transmit data for analysis.

A processing system generates distinct geometrical representations for video and audio rendering to optimize virtual environment simulation.

A high frequency decoding apparatus transforms a low frequency spectrum using an excitation class to generate a high frequency excitation spectrum.

A musical tone signal processing apparatus shifts output directions of extraction signals to expand or contract an acoustic image within defined direction ranges.

Inter-channel analysis derives time difference and amplitude ratio to generate a monaural signal that preserves waveform accuracy for efficient speech coding.

Ear tag microphones transmit audio data to a control unit that uses Independent Component Analysis algorithms to separate individual animal calls.

A mobile system estimates urine flow rate and voided volume by analyzing acoustic signals captured during urination.

An adaptive post-filter controls long-term filtering by measuring smoothed periodicity, reducing distortion at segment boundaries.

Computing device identifies playback area and listener position to determine coordinate calculation standard for HOA sound source rendering.

Encoder applies complex power ratios to maintain second-order statistics, reducing bitrate while preserving audio fidelity.

Delimiter-separated verification strings reduce coarticulation effects in speaker recognition systems.

An acoustic echo canceller estimates room response to calculate reverberation time for audio signals.

An integrated monitoring unit detects open doors and speech patterns to prompt occupants to close the entry, preventing unauthorized audio leakage.

A multistage coding process determines optimal channel pairs and weighting factors to generate weighted difference signals for compression.

Automated infant cry analyzer extracts acoustic features like fundamental frequency using cepstrum analysis and signal-to-noise ratio testing.

A simplification unit converts diverse audio signals into standardized formats for an Immersive Voice and Audio Services codec.

In-band metadata buffering enables seamless switching between channel and object audio renderers, eliminating latency-induced muting.

Residual quantization stages refine signal values to utilize unused bits and reduce quantization error.

A mask generation device extracts sound pressure information from a spectrogram and applies binarization to create an event mask.

An encoder calculates adjustment parameters based on sub-band energy and sample group positions with maximum amplitudes.

Adjusts Joint Object Coding gains using rendered presentation differences to fix level errors in spatial audio playback.

Intelligent training dictionaries capture multichannel acoustic data to enhance machine learning convergence.

Time-domain amplitude modulation inserts identifying artifacts into lossy carrier data, enabling real-time verification without audible degradation.

A distortion correcting unit suppresses echo signals using dynamic gain adjustments based on reference signal power levels.

Acoustic analysis of captured playback sound identifies active audio output type, replacing unreliable jack detection methods.

A wind noise module detects interference using log-power ratio covariance and spectral features to modify microphone signals.

Unified metadata structure merges multiple sets via reference encoding, reducing data rate while maintaining decoder compatibility.