Speech Analysis TRIZ Cases

Wavelet-based clustering and psychohaptic modeling compress multichannel vibrotactile signals while preserving perceptual detail.

Interleaved coefficient coding adds transform length switching to audio codecs, preserving backward compatibility and improving transient quality.

Closed-loop switching between delta and sparse coding cuts bits for spectral peak positions across periodic and sparse audio segments.

Redundancy bits let the decoder target corrupted payload regions and switch between full or partial concealment to reduce audio artefacts.

Metadata-guided eSBR switches between spectral translation and harmonic transposition to rebuild high frequencies without breaking legacy decoders.

Separating tonal and non-tonal bands improves wideband audio coding quality, cutting ringing noise while keeping bit rate efficient.

A common time warp across neighboring audio frames enables transform coding with adapted windows, cutting border artifacts and bit-rate overhead.

Smoothing energy depressions in decoded low-range audio improves SBR high-range generation and prevents audible degradation.

Sound-feature analysis and neural inference tailor WDRC parameters to different audio signals, improving sound quality without manual tuning.

Adaptive switching between delta and sparse coding cuts bits for spectral peak positions across periodic and sparse audio segments.

Separate dynamic range and clipping-prevention gain metadata let decoders avoid peak limiters, cutting delay and complexity.

Residual error from reconstructed low-frequency content is merged or separately encoded to improve haptic signal accuracy and synchronization.

Precomputed gain metadata shifts clipping prevention and dynamic range control toward the encoder, reducing decoder delay and computational load.

Precomputed metadata gain sequences let an audio decoder combine dynamic range compression with clipping prevention while reducing delay and complexity.

Switching between spectral translation and harmonic transposition improves musical highband reconstruction while preserving compression efficiency.

Selective concealment above a frequency threshold helps an LC3 decoder recover tonal and partial frame losses with fewer audio artefacts.

During packet loss, TCX LTP gain is faded to zero and feedback is decoupled to preserve spectral shape and avoid tonal artifacts.

Separate resolutions for diffuseness and direction parameters cut 3D audio bit-rate while preserving spatial rendering quality.

Split DRC profiles across audio frames so decoders can use a default or mode-specific profile with lower bitstream overhead.

Different time and frequency resolutions for diffuseness and direction parameters cut 3D audio bit-rate while preserving spatial quality.

Absolute phase compensation rotates the lower-energy channel more strongly, cutting stereo encoding complexity and bitrate while preserving reconstruction.

Multiple local-oscillator offsets measure IQ amplitude and phase errors, enabling filter correction that improves speaker-drive signal purity.

Signal-adaptive frequency regeneration analyzes decoded audio and attenuates tonal crossover regions to cut dissonance, beating, and ringing.

Special frames carrying pre-roll audio data let the decoder initialize instantly and switch codec configurations without audible glitches.

Band-wise spectral weighting preserves energy and avoids stereo-to-mono phase cancellation with low delay for mobile audio codecs.

Codeword-aligned audio frame partitioning protects critical data first, cutting length overhead and enabling earlier error detection and concealment.

Rotating and reordering direction parameters before quantization preserves spatial coherence and improves multi-channel audio rendering.

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

Adaptive quantization and entropy coding cut 3D audio metadata bit-rate while preserving directional quality across diffuseness levels.

Context mapping resamples prior coefficient contexts across transform lengths, preserving audio coding efficiency during frequent resolution switching.

Mode-specific coloration embeds FEC selection into coded bits, avoiding separate signaling while improving decoder mode detection.

Switching between M/S and L/R audio formats based on decoder capability preserves encoding performance and avoids codec mismatch.

Adaptive spectral-line concealment preserves tonality and spectral shape during partial or full LC3 audio frame loss.

Distortion-based scheme selection switches azimuth and elevation quantization to balance bitrate efficiency with spatial audio quality.

Context mapping reuses prior coefficient context across time or frequency resolution switches to preserve audio coding efficiency.

Bit-reversed unary transformation reshapes small-value integer distributions so Golomb coding uses fewer average bits on biased sequences.

Bijective integer transformation adapts non-Laplacian sequences for Golomb-Rice coding, cutting memory and processing overhead.

Edge-based voice biometrics isolate target speakers from noisy mixed audio, cutting latency, bandwidth use, and post-processing.

Embedding 3D audio metadata in ancillary bitstream data preserves AAC and MP3 compatibility while enabling spatial sound reproduction.

Aligns AAC information-unit borders to codeword boundaries so critical audio data can be protected and decoded with lower complexity.

Dynamic bit allocation switches between codebook and bit-difference encoding to cut multi-rate vector quantization overhead in audio coding.

IGF fills spectral gaps in the decoder domain to preserve high-frequency detail at low bitrates while reducing echo artifacts and processing load.

Embedded downmix loudness metadata lets decoders adjust gains by playback setup, preserving loudness consistency, audibility, and channel balance.

Multiple AI encoders are matched to current data using statistical parameters to avoid distortion and maintain transmission quality.

Non-uniform azimuth steps tied to elevation improve spherical direction coverage and spatial metadata accuracy for immersive audio encoding.

Adaptive switching between spectral translation and harmonic transposition improves high-band audio reconstruction for low-crossover music.

Shaping the whitened audio spectrum boosts low-energy bands to preserve spectral lines and improve encoding and decoding quality.

Selective noise filling reconstructs missing spectral coefficients in unsaturated sub-bands to improve low-bitrate decoding quality.

Interleaved frequency-domain coefficients let audio codecs add transform length switching for transients while preserving backward compatibility.

Quantized delta PCM with block headers cuts decompression compute and memory while keeping audio playback practical on IoT devices.

Encoder-generated gain metadata guides dynamic range control and clipping prevention while reducing decoder peak-limiter complexity.

Gradual filter parameter updates reduce audible artifacts during bandwidth, center frequency, and gain changes while keeping playback uninterrupted.

Bounded side and residual gains simplify stereo parameter quantization, cutting bitrate and encoding complexity while preserving spatial cues.

Fractional bit allocation per sample improves low-bitrate integer sequence compression by reversibly converting selected values while reducing decoded distortion.

Absolute phase compensation rotates lower-energy channels to simplify stereo parameter quantization while preserving spatial reconstruction at low bitrates.

Different time and frequency resolutions for direction and diffuseness parameters reduce spatial audio bit-rate while preserving immersive quality.

Short codes for zero runs and small non-zero integers cut average bit length in heavily zero-biased sequences.

Embedded special frames carry preceding encoded samples so the decoder can initialize during codec changes without interrupting playback.

Different quantization for diffuseness and direction parameters lowers 3D audio metadata bit rate while preserving spatial coding quality.

Two pitch lag estimators and correlation-based selection stabilize LTPF pitch contours for complex audio without raising complexity.

Joint MSB coding with selective spectral LSB decoding preserves audio quality while keeping bitrate and decoding complexity under control.

Adaptive VBAP triangulation avoids horizontal-plane crossings to improve 3D sound localization and reduce comb filtering in uneven speaker layouts.

Regenerates high-frequency stereo spectral gaps in the same domain, preserving timbre at low bitrates with lower compute and memory use.

Frame-by-frame coding switches by high-frequency energy and spectral shape to improve compression and reduce artificial-sounding audio.

Subvector tables and dual pointers reconstruct basis code vectors, cutting codec memory use while preserving coding efficiency.

Selective bandwidth control for FD coding tools avoids spectral holes and artificial noise while preserving full-band audio decoding.

Higher-resolution side information refines SAOC un-mixing to cut halo artifacts and inter-object crosstalk while staying backward compatible.

IGF extends high-frequency audio in the core spectral domain to preserve timbre and detail at low bitrates while reducing echo artifacts.

Periodic combined envelopes improve pitch peak approximation in audio coding while keeping variable-length code amounts small.

Centroid-sorted codebook classes narrow the search space for multiple input vectors, cutting quantization operations without losing accuracy.

Interleaved frequency-domain coefficients let audio codecs switch transform lengths for transients, reducing pre-echo while preserving legacy decoding.

Adaptive bit-length control in PVQ shape search preserves SNR and limits DSP complexity during high-rate speech and audio coding.

Encoding sine wave start positions separately from SBR noise boundaries improves high-frequency reconstruction and audio quality.

Reserved bitstream space carries loudness-state and program-boundary metadata so decoders avoid redundant processing and keep audio quality consistent.

Entropy coding adapts spectral context distance to pitch, harmonic spacing, and formants to raise audio coding efficiency with limited complexity.

Frequency-domain IGF fills spectral gaps in two-channel audio decoding, improving low-bitrate quality while reducing artifacts and overhead.

Energy and channel coherence guide which reverberation subbands to encode, reducing bit overhead while preserving restored multi-channel audio quality.

High-frequency noise fill and cross-fade smooth wide-band to narrow-band codec switching, reducing audible discontinuities without waiting for silence.

Adaptive handling of all-zero stereo frames preserves coding statistics and improves low-bitrate audio encoding efficiency.

Variable group delay filtering cuts audio path latency while reducing out-of-band noise and preserving dynamic range in personal devices.

Adaptive quantization and escape coding improve ADPCM bit rate efficiency and noise reduction across changing audio dynamic ranges.

Low and high game audio frequencies are boosted while midrange voice and music are reduced to make footsteps and explosions more prominent.

Selected spectral, pitch, and energy parameters help decoders reconstruct lost or corrupted audio frames without full redundant copies.

Multiple DRC profiles let an audio decoder match output reference level and loudness across rendering modes while limiting distortion.

Leading-sign extraction lets PVQ encode 33-bit indices on signed 32-bit DSPs, reducing dynamic range and de-indexing overhead.

Selectable group delay filtering reduces audio latency while suppressing out-of-band noise and limiting power dissipation.

A distribution identifier compresses multi-pulse algebraic codebook encoding, cutting bit use and simplifying pulse order reconstruction.

Shared noise level tracing keeps comfort noise consistent during packet-loss concealment while lowering switched audio decoder complexity.

Precomputed downmix loudness offsets let decoders match playback levels across speaker setups while preserving spatial balance and avoiding clipping.

Context mapping reuses prior coefficient data across transform and resolution changes, avoiding resets that reduce audio coding efficiency.

Reliability-weighted pitch lags and fractional pulse resynchronization improve ACELP frame-loss concealment and speech reconstruction quality.

Non-recursive input vector segmentation with recursive energy-difference coding stabilizes gain quantization and improves bit allocation.

During corrupted or missing audio frames, a two-stage fade through white noise preserves comfort noise quality and masks spectral mismatch.

A special frame embeds preceding encoded samples so the decoder starts instantly and switches codec configurations without playback glitches.

Metadata for sending frequency, sampling frequency, and their ratio preserves correct playback when compressed multichannel audio is sent at high data rates.

Independent patching and spectral processing domains let audio encoders switch SBR algorithms by signal portion for better quality and lower complexity.

Selective aliasing cancellation during codec switching cuts delay and bit overhead in hybrid audio encoding for real-time communication.

Uses shared excitation-domain noise tracing and TCX LTP fade-out to smooth audio recovery during burst packet loss.

Frame-level switching by high-frequency energy helps audio coding stay efficient across signal changes while reducing artificial sound.

Guided envelope shaping improves multi-channel transient playback by matching reconstructed channel envelopes while keeping control data low.