Speech Analysis TRIZ Cases

A two-microphone system analyzes sound signals in the frequency domain to capture three-dimensional acoustic fields.

Combining independent voice activity detectors via logical operations suppresses false alarms from babble noise while preventing speech clipping.

A multimedia ring-back system adapts caller media specifications to enable content delivery during voice call establishment.

Audio signal processing apparatus corrects digital samples using half-wave detection and coefficient tables.

A signal processing device segments reverb information into space and object components to generate audio object signals.

An emotion analysis system quantifies participant feelings to notify specific targets when levels exceed thresholds.

A decoding apparatus transforms encoded audio signals into time-domain blocks for efficient synthesis.

Discrete code token representation resolves the contradiction between high fidelity and real-world reliability in noisy environments.

Notch filtering modifies audio channels to encode embedded data without altering human perception.

Encoding multi-channel audio with spatial positioning vectors enables HOA soundfield generation for arbitrary speaker configurations.

A dual microphone electronic device converts audio signals into frequency components for processing.

A simulated call center network environment generates dynamic call scores to route traffic across multiple centers without disrupting live operations.

A postfilter control mechanism adjusts parameters based on signal stationarity to enhance reconstructed speech quality.

A multi-channel autoregressive model restores corrupted audio signals by leveraging correlated data across multiple channels.

A signal processing apparatus selects optimal decoding schemes based on input characteristics to optimize coding quality.

A noise suppression device selects the maximum power spectrum in each subband as a representative component for processing.

A multi-channel audio processing system aggregates input signals using channel clustering to support speaker identification and diarization tasks.

Dynamic buffer sizing resolves the trade-off between network jitter compensation and end-to-end delay in packet-switched networks.

Distributed injector nodes produce obfuscating sounds to mask audible messages in open listening spaces.

Processor adjusts audio signals using acoustic propagation data from a microphone array to synchronize with visual objects in augmented reality environments.

A spatial audio encoder transforms multi-channel signals into downmixes and metadata using selective parameter encoding.

Encoder quantizes low-frequency auto-regressive coefficients and derives high-frequency parts via spectral mirroring.

Autocorrelation conversion bridges narrowband and wideband LSP parameters, resolving accuracy-speed trade-offs in scalable encoding.

A locking mechanism uses Bluetooth signals and voice commands to verify user identity before changing state.

A noise estimation apparatus calculates signal variance using weighted log likelihoods and posterior probabilities across speech segments.

Segmented mixers enable selective audio filtering and private external communication without disrupting the main conference group.

Adjusts temporal gain parameters based on high-band signal characteristics to limit variability in gain shape.

Microphone arrays detect mixed audio signals to locate terminal devices, resolving rear seat accessibility constraints for on-board equipment.

An authentication apparatus calculates target feature quantities by combining air conduction and bone conduction sound features for identity verification.

Extracting extension type identifiers enables precise gain control of multi-channel objects, preventing sound quality distortion during downmix adjustments.

Information processing apparatus acquires work and emotion data to generate aptitude profiles for personnel.

Multi-order optimization minimizes spatial error across combined Ambisonics orders using weighted cost functions, improving soundfield quality for gaming.

Modifying phase differences in frequency bins encodes timing data, enabling precise audio-video synchronization without degrading signal integrity.

Segments audio frames to selectively reduce LSB depth, resolving bit rate constraints without altering original files.

An active noise control device detects actuator impedance peak frequencies to adjust control signals for consistent acoustic cancellation.

A programmable delay buffer and adaptive filter measure audio latency by matching output signals.

A speech analyzer detects pitch frequency by converting voice signals into autocorrelation waveforms for precise estimation.

A voice denoising device filters environmental noise using an interference signal before matching remaining signals against a sample library.

A noise estimation method transforms speech signals to the frequency domain and applies a non-linear gain function to enhance audio quality.

Lossy compression of residual signals minimizes upmix artifacts while reducing bit-rate requirements in object-based audio systems.

An audio decoding device uses an external setting section to override stream data parameters for audio processing.

Real-time watermark embedding identifies unauthorized copy sources in conference calls while maintaining speech signal quality.

Dynamic path selection resolves the contradiction between quantization gain and reliability by adapting to rapid frequency changes in sound signals.

A method determines audio sampling frequency using highest order line spectral frequency coefficients.

Amplitude and phase calibration of omnidirectional microphones via DSP reduces speech distortion and improves signal-to-noise ratio in noisy environments.

An error concealment unit derives a damping factor from preceding audio frames to perform adaptive fade-out.

A stereo encoding method determines scaling factors using sub-band cross correlation and energy relation coefficients.

Dynamic shift estimation aligns temporally mismatched audio signals, reducing side channel energies and improving coding efficiency without sample repetition.

Audio and video sensors feed a classifier that generates care suggestions, resolving the trade-off between automation depth and recognition accuracy.