Speech Analysis TRIZ Cases

Separating tonal and non-tonal high-band components helps low-bitrate audio encoding avoid ringing noise while preserving energy accuracy.

Bandwidth extension stays in the spectral domain to fill high-frequency gaps, preserving timbre while reducing decoder complexity and memory use.

Frequency-domain order-statistic filtering equalises microphone audio to reduce response variation, noise, and biometric mismatch.

Dynamic spectral weighting uses sharpness-based sibilant detection to reduce harsh in-car speech while preserving intelligibility.

Separating early reflections from late reverberation enables signal-dependent scaling that matches full-convolution sound with lower compute.

Different DRC profile subsets are sent in separate audio frames so decoders can match rendering modes without excessive bandwidth or distortion.

Embedding loudness-state and program-boundary metadata in reserved bitstream space helps decoders avoid redundant processing and maintain compliant audio levels.

Selective MSB and LSB coding of spectral values preserves audio quality while reducing bitrate pressure and decoder complexity.

A neural network adjusts formant attenuation and amplification by monitoring loudness to preserve spectral balance and improve mixing consistency.

Half-precision floating-point transmission preserves vibration signal headroom and precision for reliable reception-side arithmetic processing.

Adaptive window overlap follows transient location to cut look-ahead delay and pre-echo noise while preserving audio coding quality.

Non-recursive vector segmentation with recursive energy-difference coding cuts combinatorial load while preserving efficient positional coding.

Multiple bitstreams from one audio frame improve packet loss resistance and preserve audio quality with compatibility for single-bitstream decoders.

Precomputed downmix loudness offsets let decoders adapt gain by speaker configuration, keeping playback loudness and spatial balance consistent.

Interleaved coefficients let audio codecs add transform lengths for transient signals while preserving backward compatibility and low-bitrate quality.

Nested codebooks let spatial audio encoders adapt direction-parameter precision to allocated bits, cutting metadata bitrate while preserving decodability.

Signal-dependent companding adjusts frame exponents for sparse and dense transients to cut quantization noise with minimal codec overhead.

Selective stereo decoding cuts decoder power by reconstructing one channel from the other when audio conditions allow.

Frame-level DRC profile selection lets decoders match each rendering mode and preserve intelligible audio without sending every profile in every frame.

Distributed DRC profiles let a decoder match rendering modes and fall back to a known default profile when a frame lacks an applicable one.

Quantization, delta coding, and variable time windows cut memory and compute needs for audio decompression on low-power devices.

Band-wise spectral weighting preserves energy and avoids phase cancellation in stereo-to-mono downmixing with low delay.

Multiple bandpass filters with different bandwidths improve masking threshold accuracy for audio encoding while limiting computational load.

Adaptive high-frequency reconstruction switches between spectral translation and harmonic transposition to improve decoded music quality.

Segment-level gain and loudness metadata let decoders keep downmixed audio intelligible, avoid clipping, and preserve channel balance.

A phoneme classifier adds look-ahead de-essing to target sibilance without compressing non-sibilant audio or adding artifacts.

Three overlapping window functions and auxiliary windowing cut pre-echo noise and look-ahead delay in transient audio or image decoding.

Look-ahead side information such as pitch lag and gain enables immediate packet loss concealment without added algorithmic delay.

Helper parameters embedded in selected audio frames let decoders reconstruct lost, corrupted, or delayed packets with better concealment at low bitrate.

Equal-length vector segments plus recursive energy-difference coding reduce positional coding complexity and improve efficiency across uneven signal energy.

Flattening low-band energy depressions before SBR frequency shifting preserves high-band shape and reduces audible distortion.

Grouped pointwise convolutions cut neural noise suppression parameters, preserving audio quality on memory- and power-limited devices.

A time-frequency mask isolates harmonic audio from percussive content to improve identification, authentication, and media classification.

Early reflections are processed separately from late reverberation, whose signal-dependent scaling preserves full-convolution spatial perception.

Precomputed DRC metadata lets the decoder apply real-time loudness compensation and dynamic range control without full program analysis.

Block-level LUFS differences reveal dynamic variance that integrated loudness misses, improving perceived loudness matching in audio playback.

Odd-even calibration maps offset tables and converging coefficients to suppress DAC errors from current-source offsets and echo-path mismatch.

Context mapping reuses previous coefficients across time or frequency resolution changes to preserve audio coding efficiency during frequent switching.

Context mapping reuses prior coefficient context across spectral resolution changes to preserve audio coding efficiency during frequent switching.

Adaptive thresholds and down-count logic detect one pulse per period, reducing false positives and improving hearing-device power estimation.

Different resolutions for diffuseness and direction parameters cut 3D audio bit-rate while preserving spatial quality in immersive transmission.

Mapped coding contexts reuse prior coefficients across resolution and transform-length switches, improving audio coding efficiency without reset.

Touch-boundary sections use higher bit rates while steady tactile states use lower ones, with cross-fade joining to cut data and delay.

Spherical quantization and streak-limited differential coding preserve azimuth and elevation accuracy while reducing frame-erasure sensitivity.

Wideband gain companding across short audio segments cuts quantization noise in quiet passages without phase distortion or lost frequency resolution.

Higher-layer voice and text context feeds back to the decoder to correct likely errors, improving accuracy while limiting extra processing.

A special frame embeds preceding encoded samples to initialize decoder state, enabling immediate playback during codec configuration changes.

Temporal smoothing, look-ahead, and linear-scale gain conversion equalize audio loudness while reducing audible noise and preserving dynamics.

Common time warping across adjacent audio frames enables overlap-add windowing that avoids audible discontinuities and lowers bit-rate overhead.

Threshold-based DRC and AGC switch compressor curves to keep volume even without making background noise audible.

Centroid-based codebook classes and adaptive search space selection reduce vector quantization complexity while preserving reconstruction accuracy.

Frequency-domain bandwidth extension improves high-band audio coding by quantizing sub-band energy and applying controlled envelopes.

An internal loudspeaker sensor and external microphone are compared to shape mic response, reducing boomy low frequencies in calls.

Spectral tonality and amplitude ratios guide expansion coefficients to preserve high-band sound quality with lower delay and memory use.

Saved parameters from prior audio frames reconstruct erased frames, reducing audible artifacts and preserving perceptual quality.

Gain-factor compensation corrects energy errors from non-conserving upmix rules and HRTF filtering, reducing spectral coloring in binaural audio.

Separate early reflections from late reverberation and scale the reverb tail by the input signal to match full-convolution perception with less computation.

Transform-domain coefficient selection compresses IQ measurement data for bursty and frequency-hopping signals while preserving low error vector magnitude.

Embedded loudness state and program boundary metadata help decoders avoid redundant processing and maintain consistent playback quality.

IGF reconstructs high-frequency spectral gaps in the core decoder domain, preserving timbre at low bitrates while reducing echoes and complexity.

Compressed sensing, auditory masking, and source separation improve ambisonic decoding accuracy while reducing spectral distortion and speaker count.

Bandwise energy values guide same-domain high-frequency reconstruction, preserving timbre and harmonic alignment at low bitrates.

By separating direct and diffuse audio, this case restores high frequencies and transients while reducing musical noise in compressed signals.

Bandwidth extension stays in the core spectral domain, using intelligent gap filling to restore high frequencies with lower complexity and memory use.

Context reset and adaptive ACELP/TCX decoding enable random access audio playback while reducing redundant bits and decoding complexity.

Separating tonal and non-tonal bands improves wideband audio coding at low bit rates by controlling high-band noise energy and ringing.

IGF fills high-frequency spectral gaps in the same domain, preserving tonal detail at low bitrates with lower decoding complexity.

Estimating linear prediction efficiency before full encoding helps choose the right audio coding scheme with lower sub-coder complexity and overhead.

Interleaved spectral coefficients enable transform-length switching for transient audio, cutting pre-echo at low bitrates while preserving legacy decoding.

Separating tonal and non-tonal high-band components preserves wideband audio quality at low bit rates while reducing ringing noise.

Time-frequency subtraction and psychoacoustic masking analysis estimate when an audio sample is audible within complex broadcast mixes.

Grouping dominant and non-dominant subbands enables more accurate bit distribution and better sound quality in voice audio coding.

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

Linear prediction and energy-ratio coding reconstruct high-frequency audio from low-band signals, cutting bit use while preserving perceived quality.

Adaptive switching between predictive and non-predictive vector quantization reduces frame-erasure artefacts in decoded audio.

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

Gain factors derived from upmix rules and HRTF filters correct binaural energy errors and reduce spectral coloring without full multichannel rendering.

Classifying saturated and unsaturated sub-bands enables targeted noise filling of missing spectral coefficients, improving low-bitrate audio decoding quality.

Weighted window shaping blends sine and KBD behavior to improve passband selectivity and stopband rejection with low computational cost.

Band-specific loudness gains limit noise amplification in hearing-aid voice compression, preserving signal-to-noise ratio and speech recognition.

Selective time- and frequency-domain treatments restore bandwidth, harmonics, transients, reverb, and phase lost in compressed audio.

Brickwall cutoff detection guides bandwidth extension and signal fill to restore lost high frequencies, transients, and reverberation.

HRTF-aware gain correction compensates non-energy-conserving upmix rules to limit binaural energy errors and spectral coloring.

Frame-based DRC metadata uses spline-interpolated gain curves to improve time resolution and reduce aliasing distortion in audio playback.

Reinitialized predictive states and overlap-add smooth audio decoder transitions from transform to predictive coding with lower delay.

Precomputed noise coefficients let the decoder fade lost audio frames into spectrally matched comfort noise with lower complexity.

Multi-band attenuation limits loudspeaker excursion and power in real time to prevent mechanical stress and audible distortion.

Overlapping excitation frames are transformed with critical-sampled MDCT to cut coding overhead while preserving smooth speech and music quality.

Balanced-energy band coding reconstructs smooth, precise audio envelopes from cumulative sums while lowering bit-rate and complexity.

Fractional pitch lag compensation reconstructs missing speech frames with better pulse alignment, lower complexity, and fewer frequency shifts.

By coding pulse distribution separately from positions, this case cuts algebraic codebook bits and redundancy while preserving pulse order.

Wideband frequency-domain gain companding cuts quantization noise and pre-echo in lossy audio coding without phase distortion.

A shared comfort noise trace keeps fade-out consistent during burst packet losses while lowering decoder complexity and tonal artifacts.

Adaptive threshold updates extract enough high-amplitude transform coefficients for correlation while reducing extension-band encoding workload.

Zero-crossing blending and threshold prediction smooth switching between ADC paths, reducing audio artifacts across varying signal levels.

A coding method selects distinct codebook search algorithms based on input signal characteristics to reduce computational load.

A coding apparatus combines quantized spectral values into unified integers via bijective transformations for efficient encoding.

A sound signal coding method extracts downmix signals and calculates subtraction gains to generate monaural codes.

A speech noise reduction method estimates posteriori and priori signal-to-noise ratios to determine a gain for pure speech extraction.

A conversation evaluation apparatus estimates speaker utterance timing to identify switch points between main speakers.