Download Acoustic Measurement Methods for Outdoor Sites: A Comparative Study
Acoustic measurements of outdoor sites require researchers to carefully consider the appropriate method to ensure reliable results. This entails the consideration of the signal-to-noise ratio (SNR), the presence of visitors as well as restrictions that are specific to the site. The present paper presents the results of an experiment conducted in a controlled environment with the aim of comparing the reliability in the presence of different types of noise of three measurement techniques: the Exponential Sine Sweep (ESS) method using a 90-second sine sweep, this same method but with the application of time averaging of multiple measurements and the Inverse Repeated Sequence (IRS). The results are presented in relation to reverberation time and clarity parameters and demonstrate that under the test conditions the ESS method when used with a long sine sweep is the most dependable in the presence of the noise disturbances studied. These findings are of relevance for the application of convolution reverb in audio postproduction.
Download Simulating Microphone Bleed and Tom-tom Resonance in Multisampled Drum Workstations
In recent years multisampled drum workstations have become increasingly popular. They offer an alternative to recording a full drum kit if a producer, engineer or amateur lacks the equipment, money, space or knowledge to produce a quality recording. These drum workstations strive for realism, often recording up to a hundred different velocity hits of the same drum, including recordings from all microphones for each drum hit and including bleed between these microphones. This paper describes research undertaken to investigate if it is possible to simulate the snare and kick drum bleed into the tom-tom microphones and the subsequent resonance of the tom-tom that is caused, with the aim of reducing the amount of audio data that needs to be stored. A listening test was performed asking participants to identify the real recording from a simulation. The results were not statistically significant to reject the hypothesis that subjects were unable to distinguish the difference between the real and simulated recordings. This suggests listeners were unable to identify the real recording in the majority of cases.
Download A Wave Digital Filter Model of the Fairchild 670 Limiter
This paper presents a circuit-based, digital model of the prized 1950’s vintage Fairchild R 670 vacuum tube limiter. The model uses a mixture of black boxes and wave digital filters, as a step toward a fully wave digital filter design. Wave digital filters provide an efficient, modular way to digitally simulate analog circuits. A novel model for the 6386 triode is introduced to simulate the active component in a wave digital filter model of the Fairchild 670’s signal amplifier. The signal amplifier is integrated with a hybrid wave digital filter/black-box sidechain amplifier model to form a complete model of the Fairchild 670. Model test results for music and pure tones are discussed, highlighting the device’s static gain characteristics and gain reduction dependent distortion. Finally, this paper discusses the model’s salient features and their implications for designing dynamics processors.
Download Harmonic Instability of Digital Soft Clipping Algorithms
In this paper several different digital soft clipping algorithms are described and analysed. It is discussed how the quality of each algorithm can be estimated. A testing methodology is devised to show the levels of nonlinearities produced as a function of the input signal amplitude. It is proposed that, while all soft clipping algorithms produce higher order nonlinearities, the instability of the produced harmonics plays a crucial role in the transparency of the effect. Existing and novel clipping algorithms are thus compared and classified based on their measured properties, including total harmonic distortion and inter-modulation distortion estimates. This paper proposes a conclusion related to the quality and properties of different algorithms.
Download Simulation of Fender Type Guitar Preamp using Approximation and State-Space Model
This paper deals with usage of approximations for simulation of more complex audio circuits. A Fender type guitar preamp was chosen as a case study. This circuit contains two tubes and thus four nonlinear functions as well as it is a parametric circuit because of an integrated tone stack. A state-space approach was used for simulation and further, precomputed solution is approximated using nonuniform cubic splines.
Download Virtual Analog Oscillator Hard Synchronisation: Fourier series and an efficient implementation
This paper investigates a number of digital methods to produce the Analog subtractive synthesis effect of ‘Hard Synchronisation.’ While the original effect is produced by an explicit waveform phase reset, other approaches are given that produce an equivalent output. In particular, based on measurements taken from a real-analog synthesizer, a comb filtering model is proposed. This description ties in with earlier work but here an explicit structure is provided. This filter-based approach is then shown to be far more computationally efficient than the synchronisation by phase reset. This efficiency is at a minor cost as it is shown that it has a minimal impact on the sonic accuracy.
Download Visualization of Signals and Algorithms in Kronos
Kronos is a visual-oriented programming language and a compiler aimed at musical signal processing tasks. Its distinctive feature is the support for functional programming idioms like closures and higher order functions in the context of high performance real time DSP. This paper examines the visual aspect of the system. The programming user interface is discussed, along with a scheme for building custom data visualization algorithms inside the system.
Download Scattering Representation of Modulated Sounds
Mel-frequency spectral coefficients (MFSCs), calculated by averaging the spectrogram along a mel-frequency scale, are used in many audio classification tasks. Their efficiency can be partly explained by their stability to deformation in a Euclidean norm. However, averaging the spectrogram loses high-frequency information. This loss is reduced by keeping the window size small, around 20 ms, which in turn prevents MFSCs from capturing largescale structures. Scattering coefficients recover part of this lost information using a cascade of wavelet decompositions and modulus operators, enabling larger window sizes. This representation is sufficiently rich to capture note attacks, amplitude and frequency modulation, as well as chord structure.
Download Audio ADC dynamic range matching by means of a DSP equalizer and dynamics processor combination
An ideal analog-to-digital interface should be an universal interface. Different analog devices, spanning from microphones up to HiFi control amplifiers, that vary in lower or higher nominal levels or in different output resistances, should be connected to the universal analog-to-digital interface in a plug-and-play manner. The user then should not be forced to make any necessary adjustments such as setting gain ranges or adjusting input resistances. The analog-to-digital interface consists of analog signal conditioning followed by an analog-to-digital converter. Currently, the available audio analog-to-digital converters are insufficient for these requirements due to a too small dynamic range. An alternative and already known conversion technique, not perfect for all intents and purposes, but highly feasible, is the multi range analog-to-digital converter that combines analog and very precise digital signal processing. The discussion of a particular class of multi range converters is the topic of this paper.
Download Uniform Noise Sequences for Nonlinear System Identification
Noise-based nonlinear system identification techniques using Hammerstein and Wiener forms have found wide application in biological system modeling, and been applied to modeling nonlinear audio processors such as the ring modulator. These methods apply noise to the system, and project the system output onto a set of orthogonal polynomials to reveal parameters of the model. Though Gaussian sequences are invariably used to drive the unknown system, it seems clear that the statistics of the input will affect the model estimate. Motivated by the limited input and output ranges supported by analog systems, in this work, the use of an input noise sequence having a uniform distribution is explored. In addition, an error measure indicating harmonic distortion modeling accuracy is introduced. Simulation results identifying Hammerstein and Wiener systems show that the uniform and Gaussian distributions perform differently, with the uniform distribution generally producing more accurate harmonic responses. Finally, uniform noise and Gaussian noise are used to model a saturating low-pass circuit similar to that of the Tube Screamer, with the uniform distribution providing a modest improvement in noise response error.