Reverberation Effect Adding Device Circuit Redundancy
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Solution Overview
Problem
Reverberation effect adding devices require numerous circuit components and data due to the need for parallel signal processing systems with FIR filters, leading to inefficiencies and useless circuit parts when impulse response coefficients are zero.
Innovation Solution
A reverberation effect adding device is designed with a single series of impulse response coefficients stored in memory, using multiple convolution means with different sampling periods and conversion means to efficiently generate reverberation data, eliminating the need for parallel systems and reducing circuit redundancy.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If parallel signal processing systems with multiple FIR filters are used, then reverberation quality is improved, but device complexity and circuit components increase significantly
Solution Approach 1:
The patent combines multiple parallel FIR filter systems into a single unified convolution system that processes audio signals through one impulse response. This merging approach maintains the reverberation quality by using a comprehensive impulse response that encompasses multiple reflection paths, while significantly reducing circuit complexity by eliminating redundant parallel filter structures and consolidating them into a single processing chain.
Solution Approach 2:
The single impulse response system is designed to be universal, handling multiple reflection paths and reverberation characteristics within one unified structure. This multi-functional approach allows the system to perform the work of multiple specialized FIR filters through a single convolution operation, reducing the need for separate circuit components while maintaining comprehensive reverberation effects.
2Productivity
If multiple parallel signal processing systems are provided, then processing capability is improved, but useless circuit components increase when impulse response coefficients are zero
Solution Approach 1:
The patent extracts and eliminates the redundant parallel processing systems from the architecture, keeping only the essential single convolution system that performs the actual processing. By removing the unnecessary parallel FIR filter structures that would contain zero coefficients, the system maintains full processing capability through the unified impulse response while eliminating the useless circuit components that would otherwise be present in parallel configurations.
Solution Approach 2:
The patent discards the redundant parallel filter structures that would create useless circuit components with zero coefficients. The functional processing capability is recovered and concentrated into the single unified convolution system, which achieves the same or better processing efficiency without the waste of non-functional circuit elements.
3Device complexity
If a single series of impulse response coefficients is used, then device complexity is reduced, but processing efficiency must be maintained through different sampling periods
Solution Approach 1:
The patent introduces dynamic sampling period adjustment into the single convolution system. By varying the sampling periods at different stages of the impulse response processing, the system maintains high processing efficiency for different portions of the reverberation signal. This dynamic approach allows the simplified single-series structure to achieve processing speeds and quality comparable to complex parallel systems, as the sampling rate is optimized for each processing stage rather than using a fixed uniform rate throughout.
Data Source
AI summary
A reverberation effect adding device comprising a first convolution circuit which in turn comprises FIR filters (80-1 to 80-4) and an adder (accumulator) (81) which adds outputs from the FIR filters, a moving average circuit (82) which receives musical sound waveform data delayed by a predetermined number of stages in the first convolution circuit and which outputs averaged second musical sound waveform data obtained by sampling at a second sampling frequency lower than the first sampling frequency, a second convolution circuit which in turn comprises FIR filters (80-5 to 80-28) which sequentially receive the second musical sound waveform data obtained by sampling at the second sampling frequency and an adder (accumulator) (83), an interpolator (84) which receives an output from the adder (83) of the second convolution circuit, calculates an interpolated value of the output value from the adder (83), and provides the output from the adder (83) and the interpolated value from the interpolator (84) sequentially at the first sampling frequency, and an adder (85) which adds the outputs from the adder (81) and the interpolator (84) and outputs a result of the addition as reverberation data.