Wavetable synthesis is a popular sound synthesis method enabling the efficient creation of musical sounds. Using sample rate conversion techniques, arbitrary musical pitches can be generated from one wavetable or from a small set of wavetables: downsampling is used for raising the pitch and upsampling for lowering it. A challenge when changing the pitch of a sampled waveform is to avoid disturbing aliasing artifacts. Besides bandlimited resampling algorithms, the use of an integrated wavetable and a differentiation of the output signal has been proposed previously by Geiger. This paper extends Geiger’s method by using several integrator and differentiator stages to improve alias-reduction. The waveform is integrated multiple times before it is stored in a wavetable. During playback, a sample rate conversion method is first applied and the output signal is then differentiated as many times as the wavetable has been integrated. The computational cost of the proposed technique is independent of the pitch-shift ratio. It is shown that the higher-order integrated wavetable technique reduces aliasing more than the first-order technique with a minor increase in computational cost. Quantization effects are analyzed and are shown to become notable at high frequencies, when several integration and differentiation stages are used.

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This paper considers the generation of resonant waveforms from a number of perspectives. Commencing with the well-known source filter model it introduces a more advantageous heterodyne interpretation. Some variations on the basic design and comparisons with previous methods are then given. An analysis on the use of three different digital music filter structures for resonance synthesis is made, followed by an example showing how timbrally rich Frequency Modulated resonant waveforms can be synthesized.

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Virtual analog synthesis requires bandlimited source signal algorithms. An efficient methodology for the task expresses the traditionally used source waveforms or their time-derivatives as a sequence of bandlimited impulses or step functions. Approximations of the ideal bandlimited functions used in these quasi-bandlimited oscillator algorithms are typically linear-phase functions. In this paper, a general nonlinear-phase approach to the task is proposed. The discussed technique transforms an analog prototype filter to a digital filter using a modified impulse invariance transformation method that enables the impulse response to be sampled with arbitrary sub-sample shifts. The resulting digital filter is a set of parallel first- and/or second-order filters that are excited with short burst-like signals that depend on the offset of the waveform discontinuities. The discussed approach is exemplified with a number of design cases, illustrating different trade-offs between good alias reduction and low computational cost.

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Physical modeling sound synthesis for systems in 3D is a computationally intensive undertaking; the number of degrees of freedom is very large, even for systems and spaces of modest physical dimensions. The recent emergence into the mainstream of highly parallel multicore hardware, such as general purpose graphical processing units (GPGPUs) has opened an avenue of approach to synthesis for such systems in a reasonable amount of time, without severe model simplification. In this context, new programming and algorithm design considerations appear, especially the ease with which a given algorithm may be parallelized. To this end finite difference time domain methods operating over regular grids are explored, with regard to an interesting and non-trivial test problem, that of the timpani drum. The timpani is chosen here because its sounding mechanism relies on the coupling between a 2D resonator and a 3D acoustic space (an internal cavity). It is also of large physical dimensions, and thus simulation is of high computational cost. A timpani model is presented, followed by a brief presentation of finite difference time domain methods, followed by a discussion of parallelization on GPGPU, and simulation results.

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This paper presents research into scanned synthesis on a multitouch screen device. This synthesis technique involves scanning a wavetable that is dynamically evolving in the manner of a massspring network. It is argued that scanned synthesis can provide a good solution to some of the issues in digital musical instrument design, and is particularly well suited to multi-touch screens. In this implementation, vibrating mass-spring networks with a variety of configurations can be created. These can be manipulated by touching, dragging and altering the orientation of the tablet. Arbitrary scanning paths can be drawn onto the structure. Several extensions to the original scanned synthesis technique are proposed, most important of which for multi-touch implementations is the freedom of the masses to move in two dimensions. An analysis of the scanned output in the case of a 1D ideal string model is given, and scanned synthesis is also discussed as being a generalisation of a number of other synthesis methods.

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This paper discusses the development of digital audio effect applications in mobile platforms. It introduces the Mobile Csound Platform (MCP) as an agile development kit for audio programming in such environments. The paper starts by exploring the basic technology employed: the Csound Application Programming Interface (API), the target systems (iOS and Android) and their support for realtime audio. CsoundObj, the fundamental class in the MCP toolkit is introduced and explored in some detail. This is followed by a discussion of its implementation in Objective-C for iOS and Java for Android. A number of application scenarios are explored and the paper concludes with a general discussion of the technology and its potential impact for audio effects development.

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In this paper, a novel chroma extraction technique called TimeDomain Chroma Extraction (TDCE) is introduced. In comparison to many other known schemes, the calculation of a time-frequency representation is unnecessary since the TDCE is a pure sample-bysample technique. It mainly consists of a pitch tracking module that is implemented with a phase-locked loop (PLL). A set of 24 bandpass filters over two octaves is designed with the F 0 output of the pitch tracker to estimate a chroma vector. To verify the performance of the TDCE, a simple chord recognition algorithm is applied to the chroma output. The experimental results show that this novel time-domain chroma extraction technique yields good results while requiring only minor complexity and thus, enables the extraction of musical features in real-time on low-cost DSP platforms.

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We present a new onset detection algorithm which operates online in real time without delay. Our method incorporates a recurrent neural network to model the sequence of onsets based solely on causal audio signal information. Comparative performance against existing state-of-the-art online and offline algorithms was evaluated using a very large database. The new method – despite being an online algorithm – shows performance only slightly short of the best existing offline methods while outperforming standard approaches.

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In this paper we present a simple yet novel technique for harmonic/percussive separation of monaural audio music signals. Under the assumption that percussive/harmonic components exhibit vertical/horizontal lines in the spectrogram, image morphological filters are applied to the spectrogram of the input signal. The structure elements of the morphological filters are chosen to accentuate regions of the spectrogram corresponding to harmonic and percussive components. The proposed method was evaluated on the SISEC 2008/2010 development data and outperformed the baseline method adopted.

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We consider a method of hiding many audio channels in one host signal. The purpose of this is to provide a ‘mix’ that incorporates information on all the channels used to produce it, thereby allowing all, or, at least some channels to be stored in the mix for later use (e.g. for re-mixing and/or archiving). After providing an overview of some recently published audio water marking schemes in the time and transform domains, we present a method that is based on using a four least significant bits scheme to embed five MP3 files into a single 16-bit host WAV file without incurring any perceptual audio distortions in either the host data or embedded files. The host WAV file is taken to be the final mix associated with the original multi-channel data before applying minimal MP3 compression (WAV to MP3 conversion), or, alternatively, an arbitrary host WAV file into which other multi-channel data in an MP3 format is hidden. The embedded information can be encrypted and/or the embedding locations randomized on a channelby-channel basis depending on the security protocol desired by the user. The method is illustrated by providing example m-code for interested readers to reproduce the results obtained to date and as a basis for further development.

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