Download Binaural Dark-Velvet-Noise Reverberator Binaural late-reverberation modeling necessitates the synthesis of frequency-dependent inter-aural coherence, a crucial aspect of spatial auditory perception. Prior studies have explored methodologies such as filtering and cross-mixing two incoherent late reverberation impulse responses to emulate the coherence observed in measured binaural late reverberation. In this study, we introduce two variants of the binaural dark-velvet-noise reverberator. The first one uses cross-mixing of two incoherent dark-velvet-noise sequences that can be generated efficiently. The second variant is a novel time-domain jitter-based approach. The methods’ accuracies are assessed through objective and subjective evaluations, revealing that both methods yield comparable performance and clear improvements over using incoherent sequences. Moreover, the advantages of the jitter-based approach over cross-mixing are highlighted by introducing a parametric width control, based on the jitter-distribution width, into the binaural dark velvet noise reverberator. The jitter-based approach can also introduce timedependent coherence modifications without additional computational cost.
Download Audio Effect Chain Estimation and Dry Signal Recovery From Multi-Effect-Processed Musical Signals In this paper we propose a method that can address a novel task, audio effect (AFX) chain estimation and dry signal recovery. AFXs are indispensable in modern sound design workflows. Sound engineers often cascade different AFXs (as an AFX chain) to achieve their desired soundscapes. Given a multi-AFX-applied solo instrument performance (wet signal), our method can automatically estimate the applied AFX chain and recover its unprocessed dry signal, while previous research only addresses one of them. The estimated chain is useful for novice engineers in learning practical usages of AFXs, and the recovered signal can be reused with a different AFX chain. To solve this task, we first develop a deep neural network model that estimates the last-applied AFX and undoes its AFX at a time. We then iteratively apply the same model to estimate the AFX chain and eventually recover the dry signal from the wet signal. Our experiments on guitar phrase recordings with various AFX chains demonstrate the validity of our method for both the AFX-chain estimation and dry signal recovery. We also confirm that the input wet signal can be reproduced by applying the estimated AFX chain to the recovered dry signal.
Download Hyper Recurrent Neural Network: Condition Mechanisms for Black-Box Audio Effect Modeling Recurrent neural networks (RNNs) have demonstrated impressive results for virtual analog modeling of audio effects. These networks process time-domain audio signals using a series of matrix multiplication and nonlinear activation functions to emulate the behavior of the target device accurately. To additionally model the effect of the knobs for an RNN-based model, existing approaches integrate control parameters by concatenating them channel-wisely with some intermediate representation of the input signal. While this method is parameter-efficient, there is room to further improve the quality of generated audio because the concatenation-based conditioning method has limited capacity in modulating signals. In this paper, we propose three novel conditioning mechanisms for RNNs, tailored for black-box virtual analog modeling. These advanced conditioning mechanisms modulate the model based on control parameters, yielding superior results to existing RNN- and CNN-based architectures across various evaluation metrics.
Download Explicit Vector Wave Digital Filter Modeling of Circuits with a Single Bipolar Junction Transistor The recently developed extension of Wave Digital Filters based on vector wave variables has broadened the class of circuits with linear two-port elements that can be modeled in a modular and explicit fashion in the Wave Digital (WD) domain. In this paper, we apply the vector definition of wave variables to nonlinear twoport elements. In particular, we present two vector WD models of a Bipolar Junction Transistor (BJT) using characteristic equations derived from an extended Ebers-Moll model. One, implicit, is based on a modified Newton-Raphson method; the other, explicit, is based on a neural network trained in the WD domain and it is shown to allow fully explicit implementation of circuits with a single BJT, which can be executed in real time.
Download Neural Audio Processing on Android Phones This study investigates the potential of real-time inference of neural audio effects on Android smartphones, marking an initial step towards bridging the gap in neural audio processing for mobile devices. Focusing exclusively on processing rather than synthesis, we explore the performance of three open-source neural models across five Android phones released between 2014 and 2022, showcasing varied capabilities due to their generational differences. Through comparative analysis utilizing two C++ inference engines (ONNX Runtime and RTNeural), we aim to evaluate the computational efficiency and timing performance of these models, considering the varying computational loads and the hardware specifics of each device. Our work contributes insights into the feasibility of implementing neural audio processing in real-time on mobile platforms, highlighting challenges and opportunities for future advancements in this rapidly evolving field.
Download Non-Iterative Numerical Simulation in Virtual Analog: A Framework Incorporating Current Trends For their low and constant computational cost, non-iterative methods for the solution of differential problems are gaining popularity
in virtual analog provided their stability properties and accuracy
level afford their use at no exaggerate temporal oversampling. At
least in some application case studies, one recent family of noniterative schemes has shown promise to outperform methods that
achieve accurate results at the cost of iterating several times while
converging to the numerical solution. Here, this family is contextualized and studied against known classes of non-iterative methods.
The results from these studies foster a more general discussion
about the possibilities, role and prospective use of non-iterative
methods in virtual analog.
Download P-RAVE: Improving RAVE through pitch conditioning and more with application to singing voice conversion In this paper, we introduce means of improving fidelity and controllability of the RAVE generative audio model by factorizing pitch and other features. We accomplish this primarily by creating a multi-band excitation signal capturing pitch and/or loudness information, and by using it to FiLM-condition the RAVE generator. To further improve fidelity when applied to a singing voice application explored here, we also consider concatenating a supervised phonetic encoding to its latent representation. An ablation analysis highlights the improved performance of our incremental improvements relative to the baseline RAVE model. As our primary enhancement involves adding a stable pitch conditioning mechanism into the RAVE model, we simply call our method P-RAVE.
Download What you hear is what you see: Audio quality from Image Quality Metrics In this study, we investigate the feasibility of utilizing stateof-the-art perceptual image metrics for evaluating audio signals by representing them as spectrograms. The encouraging outcome of the proposed approach is based on the similarity between the neural mechanisms in the auditory and visual pathways. Furthermore, we customise one of the metrics which has a psychoacoustically plausible architecture to account for the peculiarities of sound signals. We evaluate the effectiveness of our proposed metric and several baseline metrics using a music dataset, with promising results in terms of the correlation between the metrics and the perceived quality of audio as rated by human evaluators.
Download Synthesizer Sound Matching Using Audio Spectrogram Transformers Systems for synthesizer sound matching, which automatically set the parameters of a synthesizer to emulate an input sound, have the potential to make the process of synthesizer programming faster and easier for novice and experienced musicians alike, whilst also affording new means of interaction with synthesizers. Considering the enormous variety of synthesizers in the marketplace, and the complexity of many of them, general-purpose sound matching systems that function with minimal knowledge or prior assumptions about the underlying synthesis architecture are particularly desirable. With this in mind, we introduce a synthesizer sound matching model based on the Audio Spectrogram Transformer. We demonstrate the viability of this model by training on a large synthetic dataset of randomly generated samples from the popular Massive synthesizer. We show that this model can reconstruct parameters of samples generated from a set of 16 parameters, highlighting its improved fidelity relative to multi-layer perceptron and convolutional neural network baselines. We also provide audio examples demonstrating the out-of-domain model performance in emulating vocal imitations, and sounds from other synthesizers and musical instruments.
Download A Direct Microdynamics Adjusting Processor with Matching Paradigm and Differentiable Implementation In this paper, we propose a new processor capable of directly changing the microdynamics of an audio signal primarily via a single dedicated user-facing parameter. The novelty of our processor is that it has built into it a measure of relative level, a short-term signal strength measurement which is robust to changes in signal macrodynamics. Consequent dynamic range processing is signal level-independent in its nature, and attempts to directly alter its observed relative level measurements. The inclusion of such a meter within our proposed processor also gives rise to a natural solution to the dynamics matching problem, where we attempt to transfer the microdynamic characteristics of one audio recording to another by means of estimating appropriate settings for the processor. We suggest a means of providing a reasonable initial guess for processor settings, followed by an efficient iterative algorithm to refine upon our estimates. Additionally, we implement the processor as a differentiable recurrent layer and show its effectiveness when wrapped around a gradient descent optimizer within a deep learning framework. Moreover, we illustrate that the proposed processor has more favorable gradient characteristics relative to a conventional dynamic range compressor. Throughout, we consider extensions of the processor, matching algorithm, and differentiable implementation for the multiband case.