Speech Analysis TRIZ Cases

A voice print authentication system captures acoustic signals and compares them against stored profiles to verify caller identity.

Dynamic buffering estimates network jitter using accumulated playback times, reducing delay while maintaining audio fluency.

Dynamic sound field rotation with anthropometric filters resolves computational efficiency versus directional accuracy trade-offs for immersive spatial audio.

A speech correction system reconstructs distorted audio signals using machine learning analysis of facial covering types.

Adding unique watermarks to group call audio allows devices to identify co-located users and prevent echo.

A digital microphone adjusts its analog-digital converter sampling rate to capture high-frequency audio components.

Partition audio into frequency sub-bands and calibrate left and right channels with specific parameters for precise balance.

Encoding control data into digital audio units reduces interface complexity while maintaining reliable transmission of human-machine interaction signals.

A sound processing apparatus uses phase-inverted transducers to generate counter-phase acoustic signals that cancel unwanted noise through physical superposition.

Maps captured biometric data against verified templates to associate signatures with documents, preventing unauthorized access and fraud.

System extracts acoustic features to create audio fingerprints, resolving speaker identification ambiguity when multiple users share a single device.

A QMF domain SPAR filter bank generates time-domain filters to emulate MDFT behavior for multichannel audio signals.

A neural network generates speech quality scores from audio signals without requiring reference data.

Enhanced messaging protocol uses call identifiers to reduce processing delays and enable end-to-end call tracing.

A media feature extractor analyzes audio, video, and text to detect emotional states.

A band extension device scales oversampled excitation signals using energy ratios to synthesize high frequency audio components.

A work estimation system processes inaudible sound reflections and work sounds to identify tools and tasks.

A dereverberated signal estimation method calculates instantaneous frequency from short-term Fourier transforms to determine the signal phase.

An accelerometer detects device orientation to adjust microphone weighting factors in real time.

Buffering audio data packets during extended eSCO windows eliminates voice call glitches caused by handover delays.

Electronic system monitors ambient sound levels and generates alerts to guide users toward appropriate acoustic zones.

A combined innovation codebook structure segments frequency and time domain processing to improve coding efficiency.

A digital audio editing method applies inverse effects to non-selected tracks to enhance selected track prominence.

A smoothing apparatus quantizes frequency spectral values using a log spectral envelope sequence to assign fixed-length codes.

A voice activity detection system classifies coded frames directly from the bitstream using extracted parameters.

Non-linear spectrum convolution extends harmonic content beyond the original passband, eliminating aliasing artifacts and improving speech intelligibility.

Audio-controlled device detects non-speech commands to alter output audio levels, resolving speech recognition accuracy issues caused by background noise.

Analyzes audio and user data to dynamically adjust speech playback speed, resolving the trade-off between efficiency and understanding clarity.

A blind audio watermarking system embeds signals using wavelet singular value decomposition for robust detection without original content.

Frequency domain subtraction removes embedded sound recordings from mixed audio files while minimizing artifacts through adaptive filtering.

Dynamic rating thresholds adjust sensitivity based on driving conditions to reduce false alarms while detecting hazardous audio events.

A multimodal fusion module combines audio and visual embeddings to generate identity-discriminative speaker representations.

Spectrogram-based audio fingerprinting reduces storage and processing load to enable real-time advertisement detection in streaming services.

Distance-controlled exciter manipulates audio signals to simulate realistic proximity effects in spatial scenarios.

Authentication server compares audio recordings from two proximate devices to verify user presence.

An information processing system extracts keywords from conversation data and associates them with voice feature quantities.

Pitch-based frequency band segmentation applies adaptive gain to classified audio signals for improved voice communication quality.

Reversing the conventional quantization sequence preserves spectral shape accuracy for tonal signals, reducing distortion at low bit rates.

An acoustic activity monitor uses transducers and computation to identify events from sound signals without wearables.

Sound source identification apparatus filters acoustic signals using a predetermined threshold value to isolate high-quality data for processing.