Speech Analysis TRIZ Cases

A calibration system generates frequency-banded noise samples to produce correction data for microphones.

Modifies panning gains and filter coefficients via time delays on frontal height inputs to resolve front-back confusion in surround outputs.

Audio monitoring devices record noise events and context data to validate user complaints, resolving unsubstantiated reporting issues.

Segmenting functions across air and bone receivers resolves spatial complexity constraints while maintaining high reliability in noisy environments.

Up-sampling CELP excitation signals extends audio bandwidth while reducing algorithmic delays caused by sequential MDCT processing.

Multimedia server retrieves prerecorded name audio to resolve the contradiction between written communication efficiency and spoken pronunciation accuracy.

Coding apparatus selects perceptually important subbands using lattice vectors and multi-rate indexing to maintain signal quality at partial bit rates.

Piecemeal encoding processes audio segments during capture to prevent information loss from processing delays and limited storage resources.

A voiceprint recognition system extracts user characteristics to deliver targeted audio content.

Discrete latent representations reduce bit rate while maintaining speech fidelity through separate content and speaker embedding vectors.

A communication device adjusts decoded audio output timing based on network processing event reports to maintain synchronized playback.

A high-frequency interpolation device adjusts filter coefficients based on audio bandwidth to generate continuous spectrum signals.

Selecting decorrelation filters via a combination metric reduces phase artifacts from constructive interference in virtual environments.

A hearing device evaluates speech quality by analyzing articulatory and prosodic properties of input audio signals using dedicated signal processing.

A voice processing apparatus adjusts phase difference ranges to enhance target sound direction identification accuracy.

An audio encoder generates an artificial overlap signal using a first coding model to support mode transitions.

HLDA conversion matrix reduces feature vector dimensions using class-specific covariance matrices.

A speaker identification method removes channel and noise effects from audio signals before comparing averaged spectra against enrolled models.

Audio signal processing apparatus generates multi-channel information from downmix signals using object and mix data.

Detects correlation between input digital audio signals and applies gain adjustments to resolve amplitude variations caused by phase-related mixing effects.

A multichannel audio encoding method decomposes signals into frequency bands to extract directivity information for individual sound sources.

Segmented neural networks remove reverberation from single-channel audio, reducing word error rates in speech recognition systems.

A supervisor workstation displays transient graphical alert indications overlaid on agent icons to convey real-time speech analytics data.

A spectral analysis module applies targeted gain correction to decoded tonal sound signals.

A chop detection system analyzes mean power gradients within audio spectrograms to identify periodic signal gaps.

Client devices extract embedded ultrasonic audio codes to resume playback at the exact location on a different device without network connectivity.

Automated assistants enroll users through human-to-computer dialog using visual and voice profiles without graphical interfaces.

An audio upmixer dynamically adjusts diffuse signal distribution using variable mixing matrices.

A selective voice transmission system filters audio signals using stored voiceprints to isolate authorized speakers.

Front-facing camera captures user media content for biometric identity verification.

A voice detection system differentiates original and synthetic audio signals using dynamic threshold scoring.

A buffer-memory assembly stores digital audio segments to separate computing and conversion speeds.

A voice activity detection system calculates entropy from steered response power values across a multi-microphone array to identify speech frames.

A deep neural network converts omnidirectional audio signals into unidirectional outputs by learning acoustic feature differences.

Audio decoder modifies metadata using user position offsets to render dynamic sound images, maintaining immersion when users move in virtual spaces.

A frequency band expansion device generates output signals by combining input audio with filtered white noise shaped by calculated weighting coefficients.

Headrest and mirror microphones feed a digital processor to suppress in-cabin noise, improving far-end speech intelligibility.

A collaboration server estimates audio energy to selectively perform inverse MDCT decoding on relevant streams only.

Synthesizes gate and data control signals into a single multi-level output, reducing the number of integrated circuit terminals required for LCD driving.

Microphone array estimates passenger location using arrival time analysis of spoken utterances for cabin environment control.

Selective real number processing reduces arithmetic operations and circuit size while suppressing aliasing noise.

A terminal buffers audio signals and performs frequency spectrum analysis using fast Fourier transform to identify lost high-frequency components.

Marker bits in the payload format enable flexible switching between coding modes without pre-allocating extra bits, reducing bandwidth consumption.

A watermark signal applies a phase sequence to speech spectrograms for secure authentication.

Dynamic pitch delay contour adjustment resolves low bit rate trade-offs by shifting interpolation curve endpoints based on standard deviation.

A spatial audio codec unit compresses essential parameter sets to enable real-time immersive rendering.

A universal audio driver parses codec topology information to adapt to diverse hardware configurations.

An overcomplete audio coder uses MCLT-based predictive coding to transform coefficients into polar coordinates for efficient quantization.