A semi-supervised audio model learns source-specific mixing paths from isolated recordings to separate overlapping speech in reverberant rooms.
Splitting an optical input across photodiodes and multi-band TIAs improves receiver sensitivity and frequency response while reducing distortion and jitter.
Adaptive filtering predicts network quality indicators, then residue frequency-band analysis flags anomalies early for proactive traffic management.
Feedback-driven coefficient adaptation and memory-based parameter selection enable stable, lower-complexity nonlinear signal filtering.
Differential quantization compresses convolution parameters for CNN chips, cutting bandwidth use while preserving convolution accuracy.
Projection-space correlations separate audio objects from channel mixes, enabling accurate repositioning and binaural rendering across playback setups.
Truncated BRIR subband filters in the QMF domain cut binaural rendering complexity while preserving stereo sound quality for multi-channel audio.
A neuromorphic processor extracts and phase-cancels strong RF signals, enabling cleaner spectrum analysis of weaker signals.
Approximate constraining matrices cut adaptive filter complexity, while correction modules recover cumulative update errors and preserve convergence.
Partial removal of non-causal filter components cuts sound separation delay to about 10 ms while preserving online performance for hearing aids.
By splitting FIR adder bits into accurate and approximate parts, this case cuts energy use while keeping filter accuracy acceptable.
Parallel pipelined blind source separation cuts radar signal latency and resource use while improving real-time separation and tracking.
Iterative subband gain mapping updates time-domain filter coefficients in real time, preserving low latency while tracking changing audio response.
Approximate constraining cuts adaptive filter complexity, while cumulative error correction preserves convergence in parallel block updates.
Projection-space channel correlation separates mixed audio objects for flexible re-rendering across playback setups and binaural listening.
A grating-coupled optical receiver splits signals across photodiodes and TIAs by frequency to improve sensitivity, cut distortion, and reduce jitter.
A threshold-switched filter balances noise suppression and fast pressure-change response for more reliable vacuum measurement signals.
Autocorrelation-guided step-size control helps hearing aids suppress feedback quickly while avoiding tonal-signal misadaptation.
Differential coefficients and modulo arithmetic reduce decimation FIR filter power dissipation and chip area without changing frequency response.
Parallel TIAs split by frequency range improve optical receiver sensitivity and wideband response while reducing distortion and jitter.
Dynamic feedback bit precision helps biquad stages avoid quantization errors and preserve sensitive transfer function performance.
Pipelined parallel blind source separation cuts latency, memory, and compute load while enabling real-time signal parameter tracking.
Linear transformation reduces adaptive weight dimensionality, cutting misadjustment and complexity while preserving high-order optimization.
Equalizers correct cable and circuit parasitics so electrosurgical generators can estimate true load power and deliver more consistent tissue heating.
A recursive covering map adapts filter center frequency and bandwidth in real time to separate more signals with fewer misses.
Block averaging plus an IIR filter replaces costly N-tap FIR monitoring to deliver precise PFM values with faster interim updates.
By aggregating video and auxiliary data onto one micro-coaxial cable, this case reduces cable bundle volume and frees clutch barrel space.
Adaptive digital filtering cancels self-interference in full-duplex RF links in real time, avoiding complex hardware RF cancellers.
Coefficient trajectory feedback detects phase error and retunes the adaptive filter transfer function to keep noise cancellation stable in changing acoustics.
Motion-signal frequency analysis sets adaptive filter parameters to remove ECG and other biometric motion artifacts more reliably.
Coefficient trajectory analysis detects transfer-function phase error and retunes adaptive filters for stable noise cancellation in changing environments.
Real-time loudspeaker displacement feedback shifts the high-pass cutoff to preserve bass while preventing distortion and speaker damage.
Autocorrelation-guided filter adaptation separates feedback whistling from tonal input, speeding suppression while preserving sound quality.
Adaptive filtering lets network nodes transmit and receive on the same frequency at once, reducing interference, delays, and spectrum bottlenecks.
Adaptive filters and high-speed converters enable same-frequency transmit and receive, cutting network bottlenecks and duplexing delays.
A correntropy cost function makes adaptive filtering more robust to non-Gaussian impulsive noise while preserving simple weight updates.
Phase error is identified from adaptive filter coefficient changes, then corrected to keep active noise control stable in changing acoustics.
A unified pole-update adaptive notch filter adjusts frequency and bandwidth together to suppress varying howling while maintaining convergence.
Interpolated error estimation compensates ADC nonlinearity without raising sampling rate, improving linearity with lower complexity.