Download Human Inspired Auditory Source Localization This paper describes an approach for the localization of a sound source in the complete azimuth plane of an auditory scene using a movable human dummy head. A new localization approach which assumes that the sources are positioned on a circle around the listener is introduced and performs better than standard approaches for humanoid source localization like the Woodworth formula and the Freefield formula. Furthermore a localization approach based on approximated HRTFs is introduced and evaluated. Iterative variants of the algorithms enhance the localization accuracy and resolve specific localization ambiguities. In this way a localization blur of approximately three degrees is achieved which is comparable to the human localization blur. A front-back confusion allows a reliable localization of the sources in the whole azimuth plane in up to 98.43 % of the cases.
Download Realtime system for backing vocal harmonization A system for the synthesis of backing vocals by pitch shifting of a lead vocal signal is presented. The harmonization of the backing vocals is based on the chords which are retrieved from an accompanying instrument. The system operates completely autonomous without the need to provide the key of the performed song. This simplifies the handling of the harmonization effect. The system is designed to have realtime capability to be used as live sound effect.
Download Analysis and Correction of Maps Dataset Automatic music transcription (AMT) is the process of converting the original music signal into the digital music symbol. The MIDI Aligned Piano Sounds (MAPS) dataset was established in 2010 and is the most used benchmark dataset for automatic piano music transcription. In this paper, error screening is carried out through algorithm strategy, and three data annotation problems are found in ENSTDkCl, which is a subset of MAPS, usually used for algorithm evaluation: (1) there are 342 deviation errors of midi annotation; (2) there are 803 unplayed note errors; (3) there are 1613 slow starting process errors. After algorithm correction and manual confirmation, the corrected dataset is released. Finally, the better-performing Google model and our model are evaluated on the corrected dataset. The F values are 85.94% and 85.82%, respectively, and it is correspondingly improved compared with the original dataset, which proves that the correction of the dataset is meaningful.
Download Spherical Decomposition of Arbitrary Scattering Geometries for Virtual Acoustic Environments A method is proposed to encode the acoustic scattering of objects for virtual acoustic applications through a multiple-input and
multiple-output framework. The scattering is encoded as a matrix in the spherical harmonic domain, and can be re-used and
manipulated (rotated, scaled and translated) to synthesize various
sound scenes. The proposed method is applied and validated using
Boundary Element Method simulations which shows accurate results between references and synthesis. The method is compatible
with existing frameworks such as Ambisonics and image source
methods.
Download Sparse Atomic Modeling of Audio: a Review Research into sparse atomic models has recently intensified in the image and audio processing communities. While other reviews exist, we believe this paper provides a good starting point for the uninitiated reader as it concisely summarizes the state-of-the-art, and presents most of the major topics in an accessible manner. We discuss several approaches to the sparse approximation problem including various greedy algorithms, iteratively re-weighted least squares, iterative shrinkage, and Bayesian methods. We provide pseudo-code for several of the algorithms, and have released software which includes fast dictionaries and reference implementations for many of the algorithms. We discuss the relevance of the different approaches for audio applications, and include numerical comparisons. We also illustrate several audio applications of sparse atomic modeling.
Download Sound transformation by descriptor using an analytic domain In many applications of sound transformation, such as sound design, mixing, mastering, and composition the user interactively searches for appropriate parameters. However, automatic applications of sound transformation, such as mosaicing, may require choosing parameters without user intervention. When the target can be specified by its synthesis context, or by example (from features of the example), “adaptive effects” can provide such control. But there exist few general strategies for building adaptive effects from arbitrary sets of transformations and descriptor targets. In this study, we decouple the usually direct link between analysis and transformation in adaptive effects, attempting to include more diverse transformations and descriptors in adaptive transformation, if at the cost of additional complexity or difficulty. We build an analytic model of a deliberately simple transformation-descriptor (TD) domain, and show some preliminary results.
Download Graph-Based Audio Looping and Granulation In this paper we describe similarity graphs computed from timefrequency analysis as a guide for audio playback, with the aim
of extending the content of fixed recordings in creative applications. We explain the creation of the graph from the distance between spectral frames, as well as several features computed from
the graph, such as methods for onset detection, beat detection, and
cluster analysis. Several playback algorithms can be devised based
on conditional pruning of the graph using these methods. We describe examples for looping, granulation, and automatic montage.
Download A system for data-driven concatenative sound synthesis In speech synthesis, concatenative data-driven synthesis methods prevail. They use a database of recorded speech and a unit selection algorithm that selects the segments that match best the utterance to be synthesized. Transferring these ideas to musical sound synthesis allows a new method of high quality sound synthesis. Usual synthesis methods are based on a model of the sound signal. It is very difficult to build a model that would preserve the entire fine details of sound. Concatenative synthesis achieves this by using actual recordings. This data-driven approach (as opposed to a rule-based approach) takes advantage of the information contained in the many sound recordings. For example, very naturally sounding transitions can be synthesized, since unit selection is aware of the context of the database units. The C ATERPILLAR software system has been developed to allow data-driven concatenative unit selection sound synthesis. It allows high-quality instrument synthesis with high level control, explorative free synthesis from arbitrary sound databases, or resynthesis of a recording with sounds from the database. It is based on the new software-engineering concept of component-oriented software, increasing flexibility and facilitating reuse.
Download On the evaluation of perceptual similarity measures for music Several applications in the field of content-based interaction with music repositories rely on measures which estimate the perceived similarity of music. These applications include automatic genre recognition, playlist generation, and recommender systems. In this paper we study methods to evaluate the performance of such measures. We compare five measures which use only the information extracted from the audio signal and discuss how these measures can be evaluated qualitatively and quantitatively without resorting to large scale listening tests.
Download An Efficient Algorithm for Real-Time Spectrogram Inversion We present a computationally efficient real-time algorithm for constructing audio signals from spectrograms. Spectrograms consist of a time sequence of short time Fourier transform magnitude (STFTM) spectra. During the audio signal construction process, phases are derived for the individual frequency components so that the spectrogram of the constructed signal is as close as possible to the target spectrogram given real-time constraints. The algorithm is a variation of the classic Griffin and Lim [1] technique modified to be computable in real-time. We discuss the application of the algorithm to time-scale modification of audio signals such as speech and music, and performance is compared with other methods. The new algorithm generates comparable or better results with significantly less computation. The phase consistency between adjacent frames produces excellent subjective sound quality with minimal fame transition artifacts.