DSP Audio Modeling via Transducer Parameter Extraction
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Solution Overview
Problem
Existing methods for modeling musical instrument characteristics are limited in accurately capturing the frequency response and timbre of instruments, particularly in non-linear regions, and require extensive memory and analysis databases.
Innovation Solution
A system comprising a digital signal processor (DSP) and network module that analyzes testing input and output signals in the time and frequency domains to parameterize an acoustic transducer, allowing for the modeling of instrument characteristics and modification of audio signals by identifying significant changes in output levels, thereby capturing the instrument's frequency response and timbre.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If wavetable synthesis with pre-stored sound waveforms is used, then sound quality and throughput are improved, but memory space requirement increases significantly
Solution Approach 1:
The patent creates a simplified copy of the acoustic transducer's behavior through parameterized modeling rather than storing complete waveform data. By capturing essential characteristics (frequency response, distortion characteristics, compression characteristics) in compact parameter sets, the system reproduces instrument timbre without requiring large wavetable memory stores.
Solution Approach 2:
The patent transforms the complex problem of storing complete sound waveforms into managing a small set of meaningful parameters that describe the acoustic transducer's behavior. By changing from waveform-level data to parameter-level representations (frequency response curves, distortion models, compression characteristics), the system achieves high sound quality with minimal memory requirements.
2Measurement precision
If physical sound modeling with comprehensive instrument characterization is used, then timbre accuracy is improved, but device complexity and analysis database requirements increase
Solution Approach 1:
The patent extracts only the essential characteristics of the acoustic transducer from the complete instrument system. By separating and modeling only the frequency response, distortion characteristics, and compression behavior of the transducer itself (rather than the entire instrument chain), the system achieves accurate timbre reproduction with a simplified parameter set that avoids the complexity of comprehensive instrument databases.
Solution Approach 2:
The patent divides the instrument system into separable components for independent modeling. By focusing specifically on the acoustic transducer as a distinct segment with its own characteristic parameters, the system avoids the need to model and store data for the entire instrument chain, thereby reducing device complexity while maintaining timbre accuracy.
3Measurement precision
If traditional frequency response measurement methods are used, then linear region characteristics are captured, but non-linear region characteristics are missed
Solution Approach 1:
The patent implements dynamic measurement and modeling that adapts to different operating regions of the acoustic transducer. By using measurement signals that excite both linear and non-linear behaviors and by creating models that capture distortion characteristics varying with input level, the system accurately represents frequency response across the entire operating range, not just the linear region.
Solution Approach 2:
The patent ensures continuous accurate representation of transducer behavior across all operating conditions by measuring and modeling both linear and non-linear characteristics. The parameterized model continuously adapts to represent the transducer's behavior whether operating in linear or non-linear regions, providing versatile coverage without sacrificing accuracy in either regime.
Data Source
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AI summary
A system (50) for modifying an audio signal (Audo1) comprises a portable device (50) including at least one digital signal processor DSP. The DSP comprises a first module (503) and a second module (504). The first module (503) is coupled to the interface (501) to provide a reference electric musical instrument (502) with a set of input test signals (Sig9) and obtain a set of test output signals (Resp9) via the interface (501). The second module (504) is configured to perform functions including: Analysing the set of test output signals (Resp9) to obtain a set of parameters (Prm1); parameterizing an acoustic transducer (505) and a modeler (30) to model characteristics of the reference electric musical instrument (502) based on the set of parameters (Prm1); Receiving the audio signal (Audo1) from a played musical instrument (63 / 502) to modify it into a second audio signal (Audo2) using the modeler (30) and the acoustic transducer (505), wherein the first module obtains the set of parameters (Prm1) at least by identifying frequency bins where overtones occur in the test output signals (Resp9). The set of parameters comprises two profiles (Profile A, Profile B), gain characteristics, breakup value, pre-/post-amplifiers characteristics, measured beforehand via the interface, stored and made available in the cloud. Another embodiment provides a system (60) comprising a first host (61) and a second host (62). The first host measures a reference electric musical instrument (502) via a first and second modules (503, 504) and sends the obtained set of parameters (Prm1) of a corresponding physical model over the cloud; the second host (62) retrieves the set of parameters (Prm1) from the cloud and modifies via a third module (620) the audio (Sig10) input from an actual music instrument (63) in real time, to produce an output signal (Resp10) having a similar timbre as the reference electric music instrument (502).