Speech Analysis TRIZ Cases

Speaker arrangement data embedded in AAC bitstreams guides downmix decoding, preserving compatibility while improving realistic multichannel sound.

Frame-level switching between spectral envelope and energy-differential coding improves efficiency across signal types and reduces artificial sound.

Adaptive switching between frequency- and time-domain concealment reduces modulation noise and preserves audio quality during long error intervals.

Four-part asymmetric weighting windows cut audio coding delay while keeping weights within 0-1 for quality and simpler implementation.

Adaptive window overlap follows transient location to cut encoder look-ahead and reduce pre-echo noise in audio coding.

CLD-based spatial cue rendering separates decoding from position control, enabling flexible placement of multi-object or multichannel audio.

Truncated asymmetric window overlaps enable audio transform size switching with low coder delay while preserving reconstruction and coding efficiency.

Constraining a predetermined quantization value to zero shrinks multidimensional Huffman codebooks and improves bit efficiency.

Frequency tiles and band energy values rebuild high-frequency spectral gaps with better harmonic alignment and lower codec complexity.

Mapped entropy-coding contexts carry state across time and frequency resolution changes, preserving sync while avoiding efficiency loss.

Effects are applied to estimated audio objects from down-mix and parametric data, expanding manipulation range without full decoding cost.

A frequency-domain base signal and energy control scheme extends high-frequency audio bands while preserving fine structure and coding efficiency.

Microphones and loudspeakers adapt local masking sound in real time to cut speech intelligibility and avoid unnecessary office noise.

Selective header-less and header-full RTP packetization cuts overhead while preserving frame aggregation and legacy AMR-WB interoperability.

A pulse-noise mixed codebook captures both periodic and noise components in generic speech, reducing spiky output at low bit rates.

Dual-resolution side information improves SAOC object separation, reducing halo artifacts while preserving compatibility with existing decoders.

Adaptive switching between predictive and non-predictive vector quantization reduces decoder artefacts from frame erasures.

Weighted overlap-add noise injection reconstructs lost audio frames with smoother energy transitions and fewer metallic artifacts.

Gain factors from upmix rules and HRTF filter characteristics correct binaural energy errors and reduce spectral coloring artifacts.

Gain factors derived from upmix rules and HRTF filters correct energy errors and reduce spectral coloring in binaural decoding.

Speaker arrangement data embedded in AAC comments lets decoders validate channel layouts and reproduce realistic 6.1 and 7.1 audio.

Decomposes a recording into clean signal and reverb kernel, then swaps room acoustics to keep voiceovers consistent across environments.

Precomputed gain offsets and loudness metadata help decoders keep downmixed audio consistent across speaker layouts and playback conditions.

Average amplitude analysis automatically sets audio compressor thresholds, reducing manual tuning time while delivering repeatable compression quality.

IGF rebuilds missing two-channel spectral portions at low bitrates, improving harmonic alignment while cutting artifacts, memory use, and complexity.

Expanded-spectrum decoding preserves phase relationships and tonality while reducing filter complexity for high-quality sound in resource-poor settings.

Dividing frequency envelopes into sub-vectors and codebook portions reduces quantization complexity while maintaining coding quality.

Selective IPD and OPD encoding controls frame-to-frame phase changes to cut bit rate while reducing stereo playback noise.

Threshold-based pulse encoding cuts bit waste by using fewer bits for low-index pulse combinations and offsets for larger ones.

A gain factor derived from upmix rules and HRTF filters corrects binaural energy errors and reduces spectral coloring without full multichannel rendering.

Multiple patching algorithms are selected by control information to improve SBR audio quality across signal types while keeping bit rates low.

Psychoacoustic masking guides transformed audio quantization to minimize listening-domain noise and reduce bit rate in 3D sound scenes.

Frequency-wise cross-over filtering blends decoded core audio with spectral tiles to cut ringing, dissonance, and warbling at low bitrates.

Processes ADPCM-coded audio directly from quantization indices to avoid tandem decoding, cutting complexity, delay, and quality loss.

Fading a modified MDCT spectrum toward comfort noise helps conceal lost or corrupted audio frames without repetitive or metallic sounds.

By combining pulse-like and noise-like codebook entries, this case improves low-bitrate speech coding with smoother, more natural sound.

A common excitation-domain noise trace enables smoother fade-out during burst packet loss while preserving background noise character and reducing complexity.

Neighborhood N-tuple contexts and escape-code handling improve lossless audio coding efficiency while reducing memory use.

Glitch detection and threshold-based square-wave correction improve audio-interface decoding accuracy for intelligent cipher key tokens.

By splitting audio into bands and using different second-stage filter cutoffs, this circuit preserves sound pressure and avoids distortion.

A least-squares time-frequency approach extracts stereo ambience while controlling crosstalk and preserving phase relationships.