Cheung-Fat Chan

An Efficient Approach for Two-Dimensional Parameter Estimation of a Single-Tone

In this paper, parameter estimation of a two-dimensional (2-D) single damped real/complex tone in the presence of additive white Gaussian noise is addressed. By utilizing the rank-one property of the 2-D noise-free data matrix, the damping factor and frequency for each dimension are estimated in a separable manner from the principal left and right singular vectors according to an iterative weighted least squares procedure. The remaining parameters are then obtained straightforwardly using standard least squares. The biases as well as variances of the damping factor and frequency estimates are also derived, which show that they are approximately unbiased and their performance achieves Crame¿r-Rao lower bound (CRLB) at sufficiently large signal-to-noise ratio (SNR) and/or data size conditions. We refer the proposed approach to as principal-singular-vector utilization for modal analysis (PUMA) which performs estimation in a fast and accurate manner. The development and analysis can easily be adapted for a tone which is undamped in at least one dimension. Furthermore, comparative simulation results with several conventional 2-D estimators and CRLB are included to corroborate the theoretical development of the PUMA approach as well as to demonstrate its superiority.

Efficient Approach for Sinusoidal Frequency Estimation of Gapped Data

The problem of frequency estimation for noisy sinusoidal signals from multiple segments or channels, which are referred to as gapped data, is addressed. Based on linear prediction and weighted least squares techniques, an iterative relaxation-based frequency estimator is devised and analyzed. The proposed algorithm is also extended to harmonically related frequencies. Computer simulations are conducted to compare the estimation performance of the developed approach with an existing multichannel frequency estimator and Crame¿r-Rao lower bound.

Model-Based Speech Enhancement With Improved Spectral Envelope Estimation via Dynamics Tracking

In this work, we present a model-based approach to enhance noisy speech using an analysis-synthesis framework. Target speech is reconstructed with model parameters estimated from noisy observations. In particular, spectral envelope is estimated by tracking its temporal trajectories in order to improve the noise-distorted short-time spectral amplitude. Initially, we propose an analysis-synthesis framework for speech enhancement based on harmonic noise model (HNM). Acoustic parameters such as pitch, spectral envelope, and spectral gain are extracted from HNM analysis. Spectral envelope estimation is improved by tracking its line spectrum frequency trajectories through Kalman filtering. System identification of Kalman filter is achieved via a combined design of codebook mapping scheme and maximum-likelihood estimator with parallel training data. Complete system design and experimental validations are given in details. Through performance evaluation based on a study of spectrogram, objective measures and a subjective listening test, it is demonstrated that the proposed approach achieves significant improvement over conventional methods in various conditions. A distinct advantage of the proposed method is that it successfully tackles the “musical tones” problem.

Split-dimension vector quantization of Parcor coefficients for low bit rate speech coding

A novel split vector quantization (SVQ) scheme for low bit rate coding of speech signals is proposed. In this scheme, the LPC parameter vector, which is represented by Parcor coefficients, is split into small-dimension subvectors, and each subvector is sequentially quantized according to a multistage structure that resembles a segmented lattice filter. The forward and backward prediction residuals in the segmented filter are coupled across VQ stages. The quantizer in each stage operates on the principle of minimizing the forward and backward prediction error energies similar to linear predictive analysis. Simulation results show that the new split VQ scheme can achieve transparent quantization of LPC parameters at 25 b/frame. >

Block-based bandwidth extension of narrowband speech signal by using CDHMM

In this paper we present a block-based bandwidth extension system to enhance the quality of narrowband speech signal (0-4 kHz). In memoryless bandwidth extension systems, the missing high-band components are estimated from narrowband speech using the current frame only. As the narrowband-to-wideband mapping is a one-to-many problem, this memoryless system is likely to cause hissing and whistling artifacts in the reproduced speech. Our method estimates high-band components via narrowband-to-wideband state sequence mapping using a continuous density hidden Markov model (CDHMM) on a block basis. The speech block is either one word or a sequence of words in a narrowband utterance. CDHMM estimation method avoids the one-to-many property of low-band and high-band dependency. Both subjective and objective evaluations show that hissing and whistling artifacts are reduced and the spectrally extended wideband speech (0-8 kHz) is pleasant to listen.

Quality enhancement of narrowband CELP-coded speech via wideband harmonic re-synthesis

Results for improving the quality of narrowband CELP-coded speech by enhancing the pitch periodicity and by regenerating the high-band components of speech spectra are reported. Multiband excitation (MBE) analysis is applied to enhance the pitch periodicity by re-synthesizing the speech signal using a harmonic synthesizer. The high-band magnitude spectra are regenerated by matching to low-band spectra using a trained wideband spectral codebook. Information about the voiced/unvoiced (V/UV) excitation in the high-band is derived from a training procedure and recovered by using the matched low-band index. Simulation results indicate that the quality of the wideband enhanced speech is significantly improved over the narrowband CELP-coded speech.

Wideband re-synthesis of narrowband CELP-coded speech using multiband excitation model

In this paper, a method for improving the quality of narrowband CELP-coded speech is present. The approach is to reduce the hoarse voice in CELP-coded speech by enhancing the pitch periodicity in the reproduction signal and also to reduce the muffing characteristics of narrowband speech by regenerating the highband components of speech spectra from the reproduction signal. In the proposed method, multiband excitation (MBE) analysis is performed on the reproduction speech signal from a CELP decoder and the pitch periodicity is enhanced by resynthesizing the speech signal using a harmonic synthesizer according to the MBE model. The highband magnitude spectra are regenerated by matching to lowband spectra using a trained wideband spectral codebook. Information about the voiced/unvoiced (V/UV) excitation in the highband are derived from a training procedure and then stored alongside with the wideband spectral codebook so that they can be recovered by indexing to the codebook using the matched lowband index. Simulation results indicate that the quality of the wideband resynthesized speech is significantly improved over the narrowband CELP-coded speech.

Fast stochastic codebook search through the use of odd-symmetric crosscorrelation basis vectors

A fast codebook search method for code excited linear predictive (CELP) coding of speech is described. The method relies on using a vector-sum codebook where the crosscorrelation of any pair of basis vectors is odd-symmetric. Due to this odd-symmetric crosscorrelation (OSC) property the energy term of the cost function for codebook search is a constant with respect to the search, and a simple sign detection procedure is used to locate the optimum codeword with a complexity almost independent of the codebook size. An algorithm for generating a set of OSC basis vectors is described. Simulation results show that by replacing the standard VSELP codebooks with the OSC codebooks, the same coder performance can be achieved but with a much lower complexity. An OSC-CELP coder was implemented and demonstrated to achieve good quality speech at rates below 4.8 kbps.

Wideband enhancement of narrowband coded speech using MBE re-synthesis

A method for improving the quality of narrowband CELP-coded speech is presented. The approach is to reduce the hoarse quality of CELP-coded speech by enhancing the pitch periodicity in the reproduction signal and also to reduce the muffing characteristics of narrowband speech by regenerating the highband components of speech spectra from the reproduction signal. In the proposed method, multiband excitation (MBE) analysis is performed on the reproduction speech signal from a CELP decoder and the pitch periodicity is enhanced by re-synthesizing the speech signal using a harmonic synthesizer according to the MBE model. The highband magnitude spectra are regenerated by matching to lowband spectra using a trained wideband spectral codebook. Information about the voiced/unvoiced (V/UV) excitation in the highband are derived from a training procedure and then stored alongside with the wideband spectral codebook so that they can be recovered by indexing to the codebook using the matched lowband index. Simulation results indicate that the duality of the wideband resynthesized speech is significantly improved over the narrowband CELP-coded speech.

Harmonic+noise coding using improved V/UV mixing and efficient spectral quantization

This paper presents a harmonic+noise speech coder which uses an efficient spectral quantization technique and a novel voiced/unvoiced (V/UV) mixing model. The harmonic magnitudes are coded at 23 bits/frame using the magnitude response of a linear predictive coding (LPC) system. The difference between the harmonic magnitudes and the sampled magnitude response is minimized by the closed-loop approach. The V/UV mixing is modeled by a smooth function which is derived from the speech spectrum envelope based on the flatness measure. The V/UV mixing model allows noise to be added in the harmonic portion of speech spectrum so that buzzyness is reduced. The V/UV mixing information is determined from the spectral parameters available in the decoder, no bits are needed for transmitting the V/UV information. A 1.4 kbps harmonic coder is developed. The speech quality of the coder is comparable to other harmonic coders operating at higher rates.