Speech Analysis TRIZ Cases

Order-statistic filtering estimates microphone noise floors, then inverse equalization normalizes audio for more consistent voice biometric authentication.

Dynamic switching between header-less and header-full RTP payloads preserves speech quality and legacy codec interoperability with lower overhead.

Fading a modified MDCT spectrum to white noise before FDNS smooths packet-loss concealment and keeps comfort noise consistent.

During packet loss, spectral centroid and temporal predictability guide concealment tool selection to reduce artifacts in reconstructed audio.

A spherical grid and codebook index directional audio parameters more uniformly, avoiding pole clustering and improving spatial reproduction accuracy.

Adaptive comfort noise shaping matches background noise during corrupted or missing audio frames, reducing spectral mismatch artifacts and complexity.

Early reflections are processed separately while late reverb is scaled from input signal conditions to match full-convolution sound with less computation.

Side and residual gain parameters replace complex stereo cues, cutting quantization burden and bitrate while preserving perceptual spatial audio.

Algebraic bijective transforms replace large mapping tables, enabling Golomb-Rice coding with lower memory use and less computation.

Separating consecutive zeros from non-zero integers cuts average bit count and improves compression for small-value-biased sequences.

Up-sampling and down-sampling prior context lets audio coding switch time and frequency resolutions without losing entropy coding efficiency.

Adaptive bit allocation preserves high-frequency fricative components at low bit rates by exchanging low- and high-band spectrum samples.

By adjusting harmonic detection across wideband switching, the codec keeps signal type consistent and avoids decoder discontinuities.

Selective least significant bit coding preserves key spectral detail while lowering bit demand and avoiding high-frequency quality loss.

Embedded helper parameters let decoders rebuild lost or delayed audio frames with better concealment quality while keeping VoIP bitrate low.

Jointly coding spectral MSBs while selectively skipping LSBs cuts bit demand and preserves audio quality across changing bitrate budgets.

A dual fixed and adaptive mapping scheme compresses audio while preserving low-range fidelity and enabling lightweight decoding on constrained devices.

Controls CLD-based spatial cues during decoding to reposition multi-object audio signals without fixed rendering constraints.

Shifting dynamic range control nodes across adjacent frames cuts bitrate peaks while preserving reconstructed audio quality in encoded audio streams.

A controllable band-pass filter shifts spectral ranges to detect speech and noise with lower power and less hardware complexity in sensor nodes.

Frequency-domain energy ratios and inter-microphone coherence suppress reverberation and improve speech intelligibility without a clean reference.

Sparse DRC gain samples are interpolated in the time domain after audio reconstruction to avoid aliasing, pre-echoes, and coarse gain updates.

Packet loss recovery is improved by selecting concealment tools from spectral centroid and temporal predictability cues for smoother audio decoding.

Transmitted quantizer and predictor state lets an ADPCM decoder recover from lost wireless packets with low latency and fewer audio artifacts.

Bandwidth extension stays in the spectral decoder domain, using IGF to rebuild high-frequency gaps with better timbre and lower complexity.

PCA-based stereo upmixing uses DNN envelope adjustment in the time-frequency domain to reduce spatial distortion and improve audio object placement.

A decoder flag switches highband regeneration between spectral translation and harmonic transposition to improve musical audio quality while keeping AAC SBR compatibility.

Dynamic threshold smoothing uses scene-switch detection to cut perceived audio distortion while avoiding unnatural volume shifts.

Embedded DRC profiles let an audio decoder match each rendering mode or use a known fallback profile to preserve intelligibility with lower bandwidth.

Unused trailing bits in vocoder frames carry situational awareness data with sync markers, avoiding voice delay and extra channel time.

Directly feeding sensor PDM signals into leaky spiking neurons removes separate conversion hardware, cutting silicon area and power.

Reconstruction-band energy values let IGF rebuild high-frequency spectral gaps at low bitrates while preserving tonal alignment and timbre.

Adaptive threshold updates extract a target number of high-band transform coefficients with less calculation while preserving audio coding quality.

Separating tonal and non-tonal high-band components cuts ringing noise and preserves wideband voice quality at low bit rates.

Decoder-side tonal analysis shifts frequency borders and adds cross-over filtering to regenerate high frequencies with fewer warbling and ringing artifacts.

Look-ahead side information improves CELP packet loss concealment, preserving audio quality without adding algorithmic delay.

Frequency-tile reconstruction fills high-frequency spectral gaps in audio decoding, lowering bitrate and artifacts without extra domain transforms.

Pre-encoded look-ahead side information improves CELP packet loss concealment without added delay, helping preserve audio quality.

Look-ahead side information helps CELP decoders conceal lost packets, preserve codebook sync, and avoid extra algorithmic delay.

Encoder-generated DRC and guided clipping gain metadata reduces decoder peak-limiter load while keeping audio output from clipping.

Combining spectral envelope and pitch-period information improves frequency-domain audio coding accuracy around peaks with less code.

Interleaved frequency-domain coefficients enable transform-length switching for transient audio while keeping older decoders compatible.

Two pitch lag estimates are compared using correlation measures to keep audio pitch contours stable while limiting codec complexity.

Separate DRC and clipping-prevention metadata let audio decoders avoid peak-limiter delay and complexity while preserving headroom.

Residual DAB bit errors are detected by sub-band PCM prediction and concealed by replacing only corrupted samples to suppress clicks and white noise.

Metadata-encoded DRC gain curves enable time-domain audio gain updates with higher resolution, fewer MDCT artifacts, and lower decoder complexity.

Pseudo amplitude spectra guide high-band expansion to preserve tonality while cutting decoding delay and resource use in constrained audio systems.

Deep neural network envelope adjustment helps stereo upmix processing reduce spatial distortion and place audio objects more accurately.

Selective gain coding reuses periodicity-based adjustments to cut bit count and quantization distortion without extra processing.

Low-frequency sub-band envelopes are adjusted from energy and spectral cues to allocate bits more effectively and improve audio coding quality.