Equalization of Multichannel Acoustic Systems in Oversampled Subbands
Equalization of room transfer functions (RTFs) is an important topic with several applications in acoustic signal processing. RTFs are often modeled as finite-impulse response filters, characterized by orders of thousands of taps and non-minimum phase. ...
Multichannel Eigenspace Beamforming in a Reverberant Noisy Environment With Multiple Interfering Speech Signals
In many practical environments we wish to extract several desired speech signals, which are contaminated by nonstationary and stationary interfering signals. The desired signals may also be subject to distortion imposed by the acoustic room impulse ...
Adaptive Combination of Proportionate Filters for Sparse Echo Cancellation
Proportionate adaptive filters, such as those based on the improved proportionate normalized least-mean-square (IPNLMS) algorithm, have been proposed for echo cancellation as an interesting alternative to the normalized least-mean-square (NLMS) filter. ...
Blind Source Separation Based on Cumulants With Time and Frequency Non-Properties
This paper presents new results on blind separation of instantaneously mixed independent sources based on high-order statistics together with their time and frequency non-properties (i.e., the non-stationarity and non-whiteness of sources). Separation ...
Noise Reduction Algorithms in a Generalized Transform Domain
Noise reduction for speech applications is often formulated as a digital filtering problem, where the clean speech estimate is obtained by passing the noisy speech through a linear filter/transform. With such a formulation, the core issue of noise ...
Distance-Dependent Head-Related Transfer Functions Measured With High Spatial Resolution Using a Spark Gap
A measurement of head-related transfer functions (HRTFs) with high spatial resolution was carried out in this study. HRTF measurement is difficult in the proximal region because of the lack of an appropriate acoustic point source. In this paper, a ...
Robust Multiplicative Patchwork Method for Audio Watermarking
This paper presents a Multiplicative Patchwork Method (MPM) for audio watermarking. The watermark signal is embedded by selecting two subsets of the host signal features and modifying one subset multiplicatively regarding the watermark data, whereas ...
Environmental Sound Recognition With Time–Frequency Audio Features
The paper considers the task of recognizing environmental sounds for the understanding of a scene or context surrounding an audio sensor. A variety of features have been proposed for audio recognition, including the popular Mel-frequency cepstral ...
Music Structure Analysis Using a Probabilistic Fitness Measure and a Greedy Search Algorithm
This paper proposes a method for recovering the sectional form of a musical piece from an acoustic signal. The description of form consists of a segmentation of the piece into musical parts, grouping of the segments representing the same part, and ...
Integrating Articulatory Features Into HMM-Based Parametric Speech Synthesis
This paper presents an investigation into ways of integrating articulatory features into hidden Markov model (HMM)-based parametric speech synthesis. In broad terms, this may be achieved by estimating the joint distribution of acoustic and articulatory ...
Characterization of Healthy and Pathological Voice Through Measures Based on Nonlinear Dynamics
In this paper, we propose to quantify the quality of the recorded voice through objective nonlinear measures. Quantification of speech signal quality has been traditionally carried out with linear techniques since the classical model of voice production ...
Determining Mixing Parameters From Multispeaker Data Using Speech-Specific Information
In this paper, we propose an approach for processing multispeaker speech signals collected simultaneously using a pair of spatially separated microphones in a real room environment. Spatial separation of microphones results in a fixed time-delay of ...
Robust Speaker-Adaptive HMM-Based Text-to-Speech Synthesis
This paper describes a speaker-adaptive HMM-based speech synthesis system. The new system, called ldquoHTS-2007,rdquo employs speaker adaptation (CSMAPLR+MAP), feature-space adaptive training, mixed-gender modeling, and full-covariance modeling using ...
A Cross-Language State Sharing and Mapping Approach to Bilingual (Mandarin–English) TTS
We propose a hidden Markov model (HMM)-based bilingual (Mandarin and English) text-to-speech (TTS) system to synthesize natural speech for given bilingual text. A simple baseline system consisting of two independent monolingual HMM synthesizers is built ...