Digital Technique Network TRIZ Cases

Selective coefficient filtering in each touch-sensing channel preserves signal precision while reducing filter size and circuit complexity.

Two digital filters split interpolation and phase correction to keep signal accuracy high without increasing filter coefficient memory.

Clock frequency is set from overlap size M so FFT/IFFT filter stages match throughput needs while cutting circuit scale and power use.

Step-wise weight ordering helps adaptive filters avoid local minima and cut residual echo power while reducing mean square error.

Effective input sample selection and prebuilt filter coefficients cut sampling-rate conversion complexity between base stations with different baseband rates.

A scaled sampling kernel improves digital filter attenuation with fewer taps and enables simple pass band adjustment.

An NCO-driven coefficient lookup scheme handles non-integer sample rate conversion for timing compensation with fixed-clock hardware.

A repeating I and I+1 fixed-point pattern keeps sample rate ratios accurate, preventing phase drift and buffer overruns in digital conversion.

Combining shelving filters with tuned corner frequencies enables arbitrary audio equalization while reducing ripple and phase disruption.

Position-difference FIR interpolation converts between arbitrary sampling rates without a high-rate intermediate clock, reducing circuit scale and cost.

Error-signal noise statistics set waveform weights dynamically, improving combined signal quality when input reliability and noise levels differ.

Polyphase component selection cuts FIR decimator cascade settling time, helping output signals stabilize earlier at low sampling rates.

IIR decimation and interpolation filters paired with all-pass phase compensation cut sub-band audio delay while preserving aliasing-free reconstruction.

An asynchronous sample rate converter lets RF receivers change ADC clock frequency while keeping digital paths fixed to cut spurs and warping.

Dynamic coefficient calculation lets a decimation filter adapt cutoff and notch frequencies to sampling-rate changes without larger circuits.

Parallel raw-signal splitting and recursive integration let a CIC filter cut high sampling rates by four using standard hardware.

Non-uniform ADC sampling detects occupied frequency bands with fewer samples, reducing processing power and avoiding aliasing at high frequencies.

Parallel FIR filtering uses coefficient selection and summed partial products to support high sample rates with flexible decimation and lower multiplier clocks.

Runtime FPGA reconfiguration and frequency scaling select Pareto-optimal resource use to balance power, speed, and accuracy.

Lookup-table coefficient generation cuts memory use and hardware load for high-order FIR filters with varying cut-off frequency.

Parallel delay lines and reused coefficient sets filter polyphased data at high rates while easing memory bandwidth and resource strain.

Partial sums stored in hardware registers cut memory use and avoid slow sample access during anti-aliasing downsampling.

Block-based 2D filtering stores intermediate results in registers to cut memory reads, writes, power use, and on-chip memory demand.

An iterative transmit-receive filter synthesis approach cuts inter-symbol interference while preserving crest factor and spectral quality under short filter constraints.

Parallel FIR branches with phase-specific coefficients convert high-speed sample streams while lowering internal hardware speed demands.

A shared FIR filter and selector resample independent audio streams at different rates, cutting synchronizer hardware while keeping channels aligned.

Dynamic sample rate conversion shares one converter across audio channels, balancing playback quality, channel mixing, and system complexity.

SAR-generated notch filter coefficients cut power-line noise, response time, and inter-channel phase delay in multi-channel ADC measurement.

Asymmetric filterbank windows shift coefficient energy to cut audio delay or improve reconstruction quality without added complexity.

Combining taps with common coefficients in DSP blocks cuts filter stages, resource use, and power dissipation in IC filter implementation.

A controller identifies incoming audio sample rates and switches conversion paths to cut processing load while preserving output sound quality.

A recursive-transversal filter extracts sinusoidal burst signals with lower hardware complexity and multiplication-free digital implementation.

Even-odd coefficient splitting cuts FIR filter computations by about 25% while preserving filtering performance across filter orders.

Dynamic rate estimation and SRC coefficient generation cut codec power use while converting mixed audio streams with different sampling rates.

Dynamic filter weighting tracks output variation to shorten PLL locking time while reducing jitter and preserving signal stability.

Factoring sample rate ratios into ordered filter stages cuts multirate filter design time while enabling objective selection of effective configurations.

Corrected ratio tracking and polyphase FIR interpolation reduce distortion when mixing audio streams with independent sample rates.

A five-state multiplexer lets an IIR resonator filter deliver immediate output while adapting coefficients for biomimetic voice identification.

Uses a higher-spec filter template and relaxed magnitude bounds to cut recursive filter group delay while preserving stability and acceptable response.

Minimum-phase adaptive filtering cuts group delay in audio processing while preserving sound quality, feedback resistance, and real-time response.

Polynomial FIR coefficient evaluation cuts taps, memory use, and jitter in asynchronous digital audio sample rate conversion.

Short subband filters in a complex modulated filterbank cut time-domain processing load while preserving HRTF audio quality and reducing memory use.

A split weighting and correction scheme suppresses aliasing between critically sampled subbands without long filters, reducing time and power.

Using window-function and sine lookup tables, this case cuts storage and logic cost for real-time high-order FIR coefficient generation.

Storing consecutive coefficient differences cuts bit-width and memory area while preserving high-precision digital filter reconstruction.

Selective removal of low-value subband filter taps cuts QMF audio processing complexity while keeping HRTF listening quality nearly unchanged.

Coefficient interpolation in polyphase subfilters enables flexible sample-rate conversion with lower computation and storage demand.

Shared adders, multipliers, shift registers, and multiplexers re-sample multichannel digital signals across symbol rates with one clock.

Combining real-valued subbands into complex-valued signals cuts aliasing at low frequencies while preserving frequency selectivity with lower complexity.

A single FIR filter with multiplexing and decimation replaces multiple MIMO fading filters, cutting computation while preserving independent channel filtering.