Loudspeaker Signal Calculation Using Frequency Domain Delay
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Solution Overview
Problem
Conventional loudspeaker technologies, such as wave field synthesis, face challenges in efficiently calculating and implementing delays and filters for large numbers of loudspeakers and virtual sound sources, leading to increased computational complexity and resource requirements, especially in achieving precise spatial sound reproduction across a wide area.
Innovation Solution
A device and method utilizing a combination of forward and backward transform stages, memory access controllers, filter stages, and summing stages to efficiently calculate loudspeaker signals by transforming audio signals into the spectral domain, applying filters, and backtransforming, which allows for efficient delay implementation and reduced computational overhead through strategic use of memory access and filter modifications.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Manufacturing precision
If wave field synthesis is used to achieve natural spatial sound impression across large reproduction area, then spatial sound quality is improved, but computational complexity and resource requirements increase significantly
Solution Approach 1:
The patent divides the computation into multiple stages: forward transform of audio signals to frequency domain, separate processing of amplitude scaling and time delay operations, and backward transform to time domain. This segmentation allows each stage to be optimized independently, reducing overall computational complexity while maintaining spatial sound reproduction quality.
Solution Approach 2:
The patent pre-calculates and stores amplitude scaling factors and time delay values for each loudspeaker position before actual sound reproduction. These preliminary computations are stored in memory, allowing rapid retrieval during playback without performing complex calculations in real-time, thus reducing computational burden during actual operation.
2Measurement precision
If amplitude scaling and time delay are applied to each loudspeaker signal for accurate spatial reproduction, then spatial positioning accuracy is improved, but processing time and computational resources increase
Solution Approach 1:
The patent replaces time-domain convolution operations with frequency-domain multiplication using Fast Fourier Transform (FFT). This substitution transforms computationally intensive temporal processing into more efficient frequency-domain operations, significantly reducing processing time while maintaining spatial positioning accuracy through precise amplitude scaling and phase manipulation.
3Reliability
If multiple audio sources and reflections are reproduced simultaneously through loudspeaker array, then spatial realism is improved, but calculation effort increases with number of sources and reflections
Solution Approach 1:
The patent merges the processing of multiple audio sources by accumulating their respective amplitude scaling factors and time delay values for each loudspeaker before performing the final signal generation. This combining approach reduces redundant calculations, as the system processes grouped data rather than treating each source-reflection combination separately, thereby improving calculation efficiency while maintaining spatial realism.
Data Source
AI summary
A device for calculating loudspeaker signals for a plurality of loudspeakers while using a plurality of audio sources, an audio source including an audio signal, includes a forward transform stage for transforming each audio signal, block-by-block, to a spectral domain so as to obtain for each audio signal a plurality of temporally consecutive short-term spectra, a memory for storing a plurality of temporally consecutive short-term spectra for each audio signal, a memory access controller for accessing a specific short-term spectrum among the plurality of short-term spectra for a combination consisting of a loudspeaker and an audio signal on the basis of a delay value, a filter stage for filtering the specific short-term spectrum for the combination of the audio signal and the loudspeaker by using a filter provided for the combination of the audio signal and the loudspeaker, so that a filtered shot-term spectrum is obtained for each combination of an audio signal and a loudspeaker, a summing stage for summing up the filtered short-term spectra for a loudspeaker so as to obtain summed-up short-term spectra for each loudspeaker, and a backtransform stage for backtransforming, block-by-block, summed-up short-term spectra for the loudspeakers to a time domain so as to obtain the loudspeaker signals.