2007 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics

This paper introduces speech dereverberation based on a time-varying Gaussian source model (GSM) and investigates its behavior to provide a better perspective on solving the dereverberation problem. GSM is a generalization of the autocorrelation codebook (ACC) that has recently been shown to enable us to achieve high quality speech dereverberation with only a few seconds' observation. Based on GSM... View full abstract»

The separation of multichannel audio mixtures is often addressed by the masking approach, which consists of representing the mixture signal in the time-frequency domain and associating each time-frequency bin with a small number of active sources. Adaptive time-frequency representations can increase the disjointness of the sources compared to fixed representations. However their use has not been c... View full abstract»

A learning approach for estimating sound source distance based on binaural signals is presented. The frequency-dependent coherence between the left and right ear signals is used as the distance cue. The distance estimation is based on pre-calculated coherence profiles and an energy-weighted likelihood function. The system is evaluated with different speech samples. The accuracy is best in the fron... View full abstract»

Most real microphone-array applications require sound sources to be localized in a noisy, reverberant environment. In such conditions, the steered response power using the phase transform (SRP-PHAT) has been shown to be more robust than faster, two-stage, time-difference of arrival methods. The complication is that the SRP-PHAT space has many local extrema which has required computationally costly... View full abstract»

Particle Filter-based Source Localisation algorithms attempt to track the position of a sound source - a person speaking in a room - based on the current data from a distributed microphone array as well as all previous data up to that point. This paper introduces a multi-target methodology for acoustic source tracking. The methodology is based upon the Track Before Detect (TBD) framework. The algo... View full abstract»

It is well-known that the subspace MUltiple SIgnal Classification (MUSIC) method provides high-resolution spatial spectral estimates in narrowband signal environments. However, for broadband signals, such high-resolution methods still elude researchers. This paper proposes a broadband version of the MUSIC method using a parameterized version of the spatial correlation matrix. The proposed algorith... View full abstract»

Methods for acoustic speaker tracking attempt to localize and track the position of a sound source in a reverberant environment using the data received at an array of microphones. This problem has received significant attention over the last few years, with methods based on a particle filtering principle perhaps representing one of the most promising approaches. As a Bayesian filtering technique, ... View full abstract»

The image-source method has become a ubiquitous tool in many fields of acoustics and signal processing. A technique was recently proposed to predict the energy decay (energy-time curve) in room impulse responses simulated using the image-source model. The present paper demonstrates how this technique can be efficiently used to determine the enclosure's absorption coefficients in order to achieve a... View full abstract»

Equalization of room transfer functions (RTFs) is important in many speech and audio processing applications. It is a challenging problem because RTFs are several thousand taps long and non-minimum phase and in practice only approximate measurements of the RTFs are available. In this paper, we present a subband multichannel least squares method for equalization of RTFs which is computationally eff... View full abstract»

The presence of near-end speech and ambient noise in acoustic echo cancellation makes it necessary for the adaptive filter to introduce a variable step-size to achieve high robustness and low residual error. In this paper, an optimal bin-wise block-varying step-size for the frequency-domain adaptive filter algorithm is derived and its connection to a magnitude-squared coherence (MSC) is revealed. ... View full abstract»

Speech signals for hands-free telecommunication applications are received by one or more microphones placed at some distance from the talker. In an office environment, for example, unwanted signals such as reverberation and background noise from computers and other talkers will degrade the quality of the received signal. These unwanted components have an adverse effect upon speech processing algor... View full abstract»

Efficient combinations of coding and manipulation of audio signals in the spectral domain are often desirable in communication systems. The modified discrete cosine transform (MDCT) represents a popular spectral transform in audio coding as it leads to compact signal representations. However, as the MDCT corresponds to a critically sampled filter bank, it is in general not appropriate to directly ... View full abstract»

Low delay perceptual audio coding has recently gained wide acceptance for high quality communication. While common schemes are based on the well-known Modified Discrete Cosine Transform (MDCT) filterbank, this paper describes novel coding algorithms that, for the first time, make use of dedicated low delay filterbanks, thus achieving improved coding efficiency while maintaining or even reducing th... View full abstract»

The information content of binaural signals can be beneficial to many algorithms deployed in current digital hearing aids. However, the exchange of such signals over a wireless communication link requires transmission schemes that must fulfill demanding technical constraints. We present a distributed coding algorithm that builds on psychoacoustic principles in order to achieve this goal with low b... View full abstract»

We describe a system for localizing and separating multiple sound sources from a reverberant two-channel recording. It consists of a probabilistic model of interaural level and phase differences and an EM algorithm for finding the maximum likelihood parameters of this model. By assigning points in the interaural spectrogram probabilistically to sources with the best-fitting parameters and then est... View full abstract»

A new speech-quality metric, based on time-frequency modulation, is introduced in this paper. The metric uses a cochlear model, with the signal envelope in each frequency band converted to dB above threshold. Envelopes sampled across the frequency bands give short-time spectra that are approximated using a set of mel cepstrum coefficients. The correlation between the cepstral coefficient sequences... View full abstract»

In this paper we present an illustratory process which allows privacy-preserving transactions in the context of musical databases. In particular we address the problem of matching a piece of music audio to a service database in such a way such that the database provider will not directly observe the query, nor its result, thereby preserving the privacy of the inquirer. We formulate this process wi... View full abstract»

Signal representations in overcomplete dictionaries are considered here as an alternative to the traditional transform representations for fine-grain scalable audio coding. Such representations produce sparser decompositions and thus allow better coding efficiency than transform coding at very low bitrates. Moreover, the decomposition algorithms are intrinsically progressive, and flexible enough t... View full abstract»

This paper focuses on high-fidelity multichannel audio coding based on an enhanced adaptation of the well-known sinusoidal plus noise model (SNM). Sinusoids cannot be used per se for high-quality audio modeling because they do not represent all the audible information of a recording. The noise part has also to be treated to avoid an artificial sounding resynthesis of the audio signal. Generally, t... View full abstract»

This work demonstrates a sequential Least Squares algorithm applied to the decomposition of sounds into sines-plus-residual models. For a given basis of r distinct frequency components, the algorithm derives recursively the Least Squares estimates of the associated amplitudes and phases. While a direct calculation achieves a O(nr2) complexity the main cost of our implementation is only ... View full abstract»

The Digital Waveguide Mesh (DWM) is a numerical simulation technique that has been shown to be suitable for modelling the acoustics of enclosed spaces. Previous work considered an approach using an array of spatially distributed receivers based on sound intensity probe theory to capture spatial room impulse responses (RIRs) from the DWM. A suitable process to facilitate spatial encoding of the DWM... View full abstract»

A method for decomposition of a multicomponent nonstationary signal is presented. In this method a nonlinear adaptive structure, which was first introduced as a sinusoidal tracking algorithm, is used. This structure provides excellent (instantaneous) frequency-adaptivity and (instantaneous) amplitude-adaptivity features; These features make it an ideal IF-IA estimator for a monocomponent nonstatio... View full abstract»

This paper investigates the multidimensionality of colouration perception. Here spectral variation, a dominating physical acoustic property for colouration, and its relationship to the colouration perception was studied. A method for converting colouration specified in the bark-sone domain back to the operational magnitude-frequency domain was next formulated. A colouration space was then derived ... View full abstract»

Estimating time-difference-of-arrival (TDOA) remains a challenging task when acoustic environments are reverberant and noisy. Blind channel identification approaches for TDOA estimation explicitly model multipath reflections and have been demonstrated to be effective in dealing with reverberation. Unfortunately, existing blind channel identification algorithms are sensitive to ambient noise. This ... View full abstract»

In our earlier work [1], we employed MVDR (minimum variance distortionless response) based spectral estimation instead of modified-linear prediction method [2] in pitch modification. Here, we use the Bauer method of MVDR spectral factorization, leading to a causal inverse filter rather than a noncausal filter setup with MVDR spectral estimation [1]. Further, this is employed to obtain source (or r... View full abstract»