Speech Analysis TRIZ Cases

A processor analyzes audio signals to distinguish speech from music, applying tailored decoding and signal processing schemes to each stream.

A signal de-noising method selects spectral coefficients based on inter-frame correlation to predict and adjust decoded audio signals.

A voice biometric server generates unique voice prints from audio streams to verify customer identity during communication sessions.

Segmented LPC representations control energy levels in noisy parts while maintaining tonal spectral properties during frame loss.

Voice biometrics replaces passwords and tokens, eliminating unauthorized access risks while simplifying implementation.

Mismatched windowing reduces coding delay and computational complexity while maintaining signal quality during upmix operations.

A tuned corrective frequency mask converts time-based data to the frequency domain for precise noise cancellation.

A CELP speech coding apparatus switches between orthogonal and non-orthogonal fixed codebook searches.

A stereo audio encoder maps channel difference parameters using Golomb-Rice coding to optimize bit allocation.

A digital audio decoder discriminates echo signals by defining a false alarm zone in high-energy parts to inhibit unnecessary attenuation.

AI analyzes lip-sync and facial expressions to detect impersonation probability during interviews.

An adaptive filter estimates clean speech signals from noisy microphone inputs to enable accurate intelligibility assessment.

A hearing device processor applies digital filtering to a telecoil audio signal based on transceiver activity.

A voice data transmitting system accumulates speech sections and transmits them as files using HTTP protocols.

Splitting super-wideband audio into low and high frequency bands enables effective noise suppression without speech quality degradation.

A voice band expander shifts reduced spectral components to higher frequencies.

A signal processing system generates weighted feature vectors using frame-specific reliability metrics to enhance class identification accuracy.

Predictive interpolation adjusts target channel samples to maintain temporal alignment across audio frames.

A machine learning baby monitor uses a convolutional neural network to classify infant states from video feeds and transmit alerts to mobile devices.

A voice command authentication method calculates signal-to-noise ratios and Mel-frequency cepstral coefficients to verify user identity.

A microphone array processor determines sound source distance using reverberation characteristics to classify audio signals.

A VoIP routing system collects and tailors contextual data during call setup to deliver relevant information directly to the destination endpoint.

A signal processing apparatus generates a coefficient stream and performs folding processing to complement high frequency components in sound signals.

Coding apparatus calculates similarity parameters between frequency bands to reconstruct high-frequency spectra without collapsing harmonic structure.

Calculating frequency-domain channel signal level and phase differences preserves energy during stereo down-mixing when channels have opposite phases.

Acoustic distance verification measures speaker-microphone time lag to distinguish genuine mobile devices from emulators.

Frequency compression maps high-frequency consonant information into the available passband, preserving speech intelligibility despite bandwidth constraints.

Disables voice activity detector during initial jitter adaptation packet transmission to populate the buffer.

A biometric proof portal analyzes voice and facial metadata to verify human presence during access requests.

Dynamic gain adjustment reduces iteration count and distortion by adapting step size to bit count differences, ensuring fast convergence to optimal values.

Adaptive non-uniform tiling segments bins via cascaded transforms to reduce aliasing and lower bitrates without audible artifacts.

Natural language processing segments medical reports into information units using context-free grammar for automated classification.

Automated audio processing converts speech to text and metadata, enabling targeted video archiving that reduces manual search time.

A localization algorithm uses blind source separation to isolate target audio signals from mixed microphone inputs.

An audio enhancement network trained on demodulated radio signals processes audio segments to improve intelligibility.

DSP algorithms unfold stereo signals in time using psychoacoustic grouping to overcome limited spatial realism in conventional audio reproduction.

Virtual directional microphones calculate energy ratios to detect voice activity, resolving noise suppression accuracy limits.

Machine learning classifies user-defined sounds to trigger visual or haptic alerts on portable devices.

A watermarking architecture inserts messages into multiple spectral channels to maintain imperceptibility in speech-like audio.

Plastic slide bearings enable fore-aft seat movement within a compact anchor plate assembly, preventing rust and debris accumulation in off-road vehicles.

A pitch extraction device re-encodes bit streams to calculate fundamental frequency via autocorrelation.

Audio decoder maps dynamic objects to static speaker configurations, reducing processing complexity while maintaining immersive audio quality.

Maximum likelihood estimation of spectral variances separates direct sound from reverberation, improving speech intelligibility in hearing aids.

Precomputed relative transfer functions enable accurate direction-of-arrival estimation in noisy environments while reducing computational complexity.

A vehicle driving sound control system generates acoustic output by applying a filter derived from voice signal characteristics to default engine sounds.

A dual-microphone voice system separates voice and noise signals for stable cancellation.

A conversion entity renders voicemail messages electronically accessible via telephone network forwarding.