Download Segregation of Two Simultaneously Arriving Narrowband Noise Signals as a Function of Spatial and Frequency Separation
The present paper details a set of subjective measurements that were carried out in order to investigate the perceptual fusion and segregation of two simultaneously presented ERB-bandlimited noise samples as a function of their frequency separation and difference in the direction of arrival. This research was motivated by the desire to gain insight to virtual source technology in multichannel listening and virtual acoustics applications. The segregation threshold was measured in three different spatial configurations, namely with a 0◦ , a 22.5◦ , or a 45◦ azimuth separation between the two noise signals. The tests were arranged so that the subjects adjusted the frequency gap between the two noise bands until they in their opinion were at the threshold of hearing two separate sounds. The results indicate that the frequency separation threshold is increased above approximately 1.5 kHz. The effect of angle separation between ERB-bands was less significant. It is therefore assumed that the results can be accounted by the loss of accuracy in the neural analysis of the complex stimulus waveform fine structure. The results are also relatively divergent between subjects. This is believed to indicate that sound fusion is an individual concept and partly utilizes higher-level processing.
Download CMOS Implementation of an Adaptive Noise Canceller into a Subband Filter
In recent years the demand for mobile communication has increased rapidly. While in the early years of mobile phones battery life was one of the main concerns for developers speech quality is now becoming one of the most important factors in the development of the next generation of mobile phones. This paper describes the CMOS implementation of an adaptive noise canceller (ANC) into a subband filter. The ANC-Subband filter is able to reduce noise components of real speech without prior knowledge of the noise properties. It is predestined to be used in mobile devices and therefore, uses a very low clock frequency resulting in a small power consumption. This low power consumption combined with its small physical size enables the circuit also be used in hearing aids to efficiently reduce noise contained in the speech signal.
Download Sound Spatialization with Particle Systems
Download Audio Rendering System Design for an Object Oriented Audio Visual Human Perception Assessment Tool
The cognitive processes behind human bimodal (audio visual) perception are not well understood. This contribution presents an approach to reach a deeper understanding by means of subjective assessments of (interactive) audio visual applications. A tool developed for performing these assessments is introduced, and the audio rendering system design of that tool is explained: its modular character, the signal processing flow as well as the possible reproduction setups are discussed. Finally, an example for the assessment of geometrically based room simulation and preliminary test results are given.
Download FEAPI: a low level feature extraction plugin API
This paper presents FEAPI, an easy-to-use platform-independent plugin application programming interface (API) for the extraction of low level features from audio in PCM format in the context of music information retrieval software. The need for and advantages of using an open and well-defined plugin interface are outlined in this paper and an overview of the API itself and its usage is given.
Download Generalised Prior Subspace Analysis for Polyphonic Pitch Transcription
A reformulation of Prior Subspace Analysis (PSA) is presented, which restates the problem as that of fitting an undercomplete signal dictionary to a spectrogram. Further, a generalization of PSA is derived which allows the transcription of polyphonic pitched instruments. This involves the translation of a single frequency prior subspace of a note to approximate other notes, overcoming the problem of needing a separate basis function for each note played by an instrument. Examples are then demonstrated which show the utility of the generalised PSA algorithm for the purposes of polyphonic pitch transcription.
Download Polyphonic music analysis by signal processing and support vector machines
In this paper an original system for the analysis of harmony and polyphonic music is introduced. The system is based on signal processing and machine learning. A new multi-resolution, fast analysis method is conceived to extract time-frequency energy spectrum at the signal processing stage, while support vector machine is used as machine learning technology. Aiming at the analysis of rather general audio content, experiments are made on a huge set of recorded samples, using 19 music instruments combined together or alone, with different polyphony. Experimental results show that fundamental frequencies are detected with a remarkable success ratio and that the method can provide excellent results in general cases.
Download Improved method for extraction of partial’s parameters in polyphonic transcription of piano higher octaves
Polyphonic transcription is specially challenging in piano higher octaves due to the complexity of the spectrum of notes and therefore, chords. Besides the fundamental and second partial components, other spectral elements appears. The three peaks related to the unison as well as the second harmonic of the fundamental unison can be distinguished in most measures. Furthermore, intermodulation components are also present when non-linearity is high enough. This paper compares several methods to improve the training process that allows to synthesize the spectral patterns and masks used in transcription methods.
Download Live Tracking of Musical Performances using On-Line Time Warping
Dynamic time warping finds the optimal alignment of two time series, but it is not suitable for on-line applications because it requires complete knowledge of both series before the alignment can be computed. Further, the quadratic time and space requirements are limiting factors even for off-line systems. We present a novel on-line time warping algorithm which has linear time and space costs, and performs incremental alignment of two series as one is received in real time. This algorithm is applied to the alignment of audio signals in order to follow musical performances of arbitrary length. Each frame of audio is represented by a positive spectral difference vector, emphasising note onsets. The system was tested on various test sets, including recordings of 22 pianists playing music by Chopin, where the average alignment error was 59ms (median 20ms). We demonstrate one application of the system: the analysis and visualisation of musical expression in real time.
Download Gestural exploitation of ecological information in continuous sonic feedback – The case of balancing a rolling ball
Continuous sensory–motor loops form a topic dealt with rather rarely in experiments and applications of ecological auditory perception. Experiments with a tangible audio–visual interface around a physics-based sound synthesis core address this aspect. Initially dealing with the evaluation of a specific work of sound and interaction design, they deliver new arguments and notions for non-speech auditory display and are also to be seen in a wider context of psychoacoustic knowledge and methodology.