There is, I think, something ethereal about i —the square root of minus one. I remember first hearing about it at school. It seemed an odd beast at that time—an intruder hovering on the edge of reality.

Usually familiarity dulls this sense of the bizarre, but in the case of i it was the reverse: over the years the sense of its surreal nature intensified. It seemed that it was impossible to write mathematics that described the real world in …

I and i

In this paper, we discuss the evaluation of blind audio source separation (BASS) algorithms. Depending on the exact application, different distortions can be allowed between an estimated source and the wanted true source. We consider four different sets of such allowed distortions, from time-invariant gains to time-varying filters. In each case, we decompose the estimated source into a true source part plus error terms corresponding to interferences, additive noise, and algorithmic artifacts. Then, we derive a global performance measure using an energy ratio, plus a separate performance measure for each error term. These measures are computed and discussed on the results of several BASS problems with various difficulty levels

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Performance measurement in blind audio source separation

Speech processing algorithms are often developed demonstrating improvements over the state-of-the-art, but sometimes at the cost of high complexity. This makes algorithm reimplementations based on literature difficult, and thus reliable comparisons between published results and current work are hard to achieve. This paper presents a new collaborative and freely available repository for speech processing algorithms called COVAREP, which aims at fast and easy access to new speech processing algorithms and thus facilitating research in the field. We envisage that COVAREP will allow more reproducible research by strengthening complex implementations through shared contributions and openly available code which can be discussed, commented on and corrected by the community. Presently COVAREP contains contributions from five distinct laboratories and we encourage contributions from across the speech processing research field. In this paper, we provide an overview of the current offerings of COVAREP and also include a demonstration of the algorithms through an emotion classification experiment.

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COVAREP — A collaborative voice analysis repository for speech technologies

We aim to assess the perceived quality of estimated source signals in the context of audio source separation. These signals may involve one or more kinds of distortions, including distortion of the target source, interference from the other sources or musical noise artifacts. We propose a subjective test protocol to assess the perceived quality with respect to each kind of distortion and collect the scores of 20 subjects over 80 sounds. We then propose a family of objective measures aiming to predict these subjective scores based on the decomposition of the estimation error into several distortion components and on the use of the PEMO-Q perceptual salience measure to provide multiple features that are then combined. These measures increase correlation with subjective scores up to 0.5 compared to nonlinear mapping of individual state-of-the-art source separation measures. Finally, we released the data and code presented in this paper in a freely available toolkit called PEASS.

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Subjective and Objective Quality Assessment of Audio Source Separation

A new method for the estimation of multiple concurrent pitches in piano recordings is presented. It addresses the issue of overlapping overtones by modeling the spectral envelope of the overtones of each note with a smooth autoregressive model. For the background noise, a moving-average model is used and the combination of both tends to eliminate harmonic and sub-harmonic erroneous pitch estimations. This leads to a complete generative spectral model for simultaneous piano notes, which also explicitly includes the typical deviation from exact harmonicity in a piano overtone series. The pitch set which maximizes an approximate likelihood is selected from among a restricted number of possible pitch combinations as the one. Tests have been conducted on a large homemade database called MAPS, composed of piano recordings from a real upright piano and from high-quality samples.

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Multipitch Estimation of Piano Sounds Using a New Probabilistic Spectral Smoothness Principle

This article reports on developments conducted in the framework of the SampleOrchestrator project. We assembled for this project a set of tools allowing for the interactive real-time synthesis of automatically analysed and annotated audio files. Rather than a specific technique we present a set of components that support a variety of different interactive real-time audio processing approaches such as beatshuffling, sound morphing, and audio musaicing. We particularly insist on the design of the central element of these developments, an optimised data structure for the consistent storage of audio samples, descriptions, and annotations closely linked to the SDIF file standard.

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MuBu and Friends – Assembling Tools for Content Based Real-Time Interactive Audio Processing in Max/MSP

The article is concerned with the estimation of fundamental frequencies, or F0s, in polyphonic music. We propose a new method for jointly evaluating multiple F0 hypotheses based on three physical principles, harmonicity, spectral smoothness and synchronous amplitude evolution, within a single source. Based on the generative quasiharmonic model, a set of hypothetical partial sequences is derived and an optimal assignment of the observed peaks to the hypothetical sources and noise is performed. The hypothetical partial sequences are then evaluated by a score function which formulates the guiding principles in a mathematical manner. The algorithm has been tested on a large collection of artificially mixed polyphonic samples and the results show the competitive performance of the proposed method.

Multiple fundamental frequency estimation of polyphonic music signals

On cepstral and all-pole based spectral envelope modeling with unknown model order

In this work we address the problem of all pole spectral envelope estimation for speech signals. The currently widely used all pole spectral envelope model suffers from well-known systematic errors and more severely from model order mismatch. We will propose a procedure to first establish a band limited interpolation of the observed spectrum using a recently rediscovered true envelope estimator and then using the band limited envelope to derive an all pole envelope model named TE-LPC. The band-limited envelope that is used to derive the all pole envelope model reduces the problem of the unknown all pole model order. For the experimental investigation we propose a new perceptually motivated residual spectral peak flatness measure. The experimental results demonstrate that the proposed method significantly increases the spectral flatness for the perceptually especially important low order harmonics of voiced utterances.

/pdf/improving-lpc-spectral-envelope-extraction-of-voiced-speech-1cn7y22342.pdf

Improving Lpc Spectral Envelope Extraction Of Voiced Speech By True-Envelope Estimation

In glottal source analysis, the phase minimization criterion has already been proposed to detect excitation instants. As shown in this article, this criterion can also be used to estimate the shape parameter of a glottal model (ex. Liljencrants-Fant model) and not only its time position. Additionally, we show that the shape parameter can be estimated independently of the glottal model position. The reliability of the proposed methods is evaluated with synthetic signals and compared to that of the IAIF and minimum/maximum-phase decomposition methods. The results of the methods are evaluated according to the influence of the fundamental frequency and noise. The estimation of a glottal model is useful for the separation of the glottal source and the vocal-tract filter and therefore can be applied in voice transformation, synthesis and also in clinical context or for the study of the voice production.

Axel Röbel

Papers

MuBu and Friends – Assembling Tools for Content Based Real-Time Interactive Audio Processing in Max/MSP

Multiple fundamental frequency estimation of polyphonic music signals

On cepstral and all-pole based spectral envelope modeling with unknown model order

Improving Lpc Spectral Envelope Extraction Of Voiced Speech By True-Envelope Estimation

Phase minimization for glottal model estimation