Sangwon Kang

Low-complexity predictive trellis-coded quantization of speech line spectral frequencies

Low-complexity block-constrained trellis coded quantization (BC-TCQ) structures are studied, and a predictive BC-TCQ encoding method is developed for quantization of line spectrum frequency (LSF) parameters for speech coding applications. The performance is compared with the linear predictive coding (LPC) vector quantizers used in the AMR-WB (ITU-G.722.2) and IS-641 speech coding standards, demonstrating reduction in spectral distortion (SD) and significant reduction in encoding complexity.

Trellis excitation speech coding at low bit rates

Trellis coded vector quantization (TCVQ) and code-excited linear prediction (CELP) coding are combined to form an efficient low-bit rate speech coding system. The resulting system uses a trellis search to select the synthesis filter excitation sequence, and is referred to as trellis excitation coding (TEC). Simulations are performed for encoding rates of 6.4 and 8 kbps. Informal listening tests indicate that the 8 kbps TEC system has quality roughly between that of 6-bit and 7-bit /spl mu/-law PCM with /spl mu/=255. The 6.4 kbps TEC system provides speech quality between 5-bit and 6-bit /spl mu/-law PCM. A subjective comparison with vector sum excited linear prediction (VSELP) indicates that the 8 kbps TEC and the VSELP reconstructed speech is about equally preferable. >

Low-complexity predictive trellis coded quantization of wideband speech LSF parameters

In this paper, low-complexity block-constrained trellis coded quantization (BC-TCQ) structures are introduced, and a predictive BC-TCQ encoding method is developed for quantization of line spectrum frequencies (LSF) parameters for wideband speech coding applications. The performance is compared to the linear predictive coding (LPC) vector quantizers used in the AMR-WB (ITU-G.722.2) speech coding standard, demonstrating reduction in spectral distortion and significant reduction in encoding complexity.

Design of a variable rate algorithm for the 8 kb/s CS-ACELP coder

In this paper, we propose an efficient variable rate algorithm for G.729. The variable rate algorithm consists of two main subjects, the rate determination algorithm and the design of sub rate coders. For the rate determination, we combine the energy-thresholding method, the phonetic segmentation method and the variable hangover period method. Through the analysis of noise features, a 1 kb/s sub rate coder is designed for coding the background noise signal. Also, we design a 4 kb/s sub rate coder for the unvoiced parts. Experiments show that the proposed variable rate conjugate structure-algebraic code excited linear prediction (CS-ACELP) coder produces the same speech quality as G.729, at the average bit rate of 4.4 kb/s.

Deep neural network-based linear predictive parameter estimations for speech enhancement

This study presents a speech enhancement technique to improve noise corrupted speech via deep neural network (DNN)-based linear predictive (LP) parameter estimations of speech and noise. With regard to the LP coefficient estimation, an enhanced estimation method using a DNN with multiple layers was proposed. Excitation variances were then estimated via a maximum-likelihood scheme using observed noisy speech and estimated LP coefficients. A time-smoothed Wiener filter was further introduced to improve the enhanced speech quality. Performance was evaluated via log spectral distance, a composite multivariate adaptive regression splines modelling-based measure, and a segmental signal-to-noise ratio. The experimental results revealed that the proposed scheme outperformed competing methods.

Artificial bandwidth extension using deep neural network-based spectral envelope estimation and enhanced excitation estimation

The authors propose a robust artificial bandwidth extension (ABE) technique to improve narrowband (NB) speech signal quality using an enhanced spectrum envelope and excitation estimation. For envelope estimation, they propose an enhanced envelope estimation method using a deep neural network with multiple layers. For excitation estimation, they use a whitened NB excitation signal that is generated by passing the excitation signal through a whitening filter. An adaptive spectral double shifting method is introduced to obtain an enhanced wideband (WB) excitation signal. The proposed ABE system is applied to the decoded output of an adaptive multi-rate (AMR) codec at 12.2 kbps. They evaluate its performance using log spectral distortion, a WB perceptual evaluation of speech quality, and a formal listening test. The objective and subjective evaluations confirm that the proposed ABE system provides better speech quality than AMR at the same bit rate.

A fast-search algorithm for the predictive split VQ of LPC parameters

A fast-search method is introduced to reduce the searching range of a codebook using the ordering property of LSF parameters. We apply the proposed fast-search method to the G.723.1 coder with the predictive split VQ structure. Simulation results indicate that using this fast-search method, the average searching range of the codebook can be reduced by 20.1% without degradation of the spectral distortion (SD) and additional increases of memories. The reduction of 20.1% in the average searching range results in the decrease of 19.1%, 20.1%, 19.4%, and 12.2% in the number of additions, subtractions, multiplies, and comparisons, respectively.

Flexible spectrum coding in the 3GPP EVS codec

This paper proposes a flexible encoding technique based on multi-stage multiple scale lattice vector quantization and block-constrained trellis coded vector quantization. It is used for the spectrum encoding, more precisely encoding of the LSF parameters, and incorporated in the recently standardized 3GPP EVS codec. The proposed method can handle multiple bit allocations and signal types with low complexity and low memory requirements.

Bandwidth extension of G.729 speech coder using search-free codebook mapping

A search-free codebook mapping algorithm for bandwidth extension in G.729 compressed domain is introduced in this paper. We design a wideband line spectrum pair (LSP) codebook which is coupled with the same index as the LSP codebook of the G.729 speech codec. The received narrowband LSP codebook indices are used to directly induce wideband LSP codewords. Thus, the proposed scheme eliminates codebook search processing to estimate the wideband spectrum envelope. Its performance was assessed via the perceptual evaluation of speech quality (PESQ), spectral distortion (SD), informal listening tests, and weighted million operations per second (WMOPS) calculations.

Designing Algebraic Trellis Code as a New Fixed Codebook Module for ACELP Coder

In this paper, a block-constrained trellis coded quantization (BC-TCQ) algorithm is combined with an algebraic codebook to produce an algebraic trellis code (ATC) to be used in ACELP coding. In ATC, the set of allowed algebraic codebook pulse positions is expanded, and the expanded set is partitioned into subsets of pulse positions; the trellis branches are labeled with these subsets. The list Viterbi algorithm (LVA) is used to select the excitation codevector. The combination of an ATC codebook and LVA trellis search algorithm is denoted as an ATC-LVA block code. The ATC-LVA block code is used as the fixed codebook of the AMRWB 8.85kbps mode, reducing complexity compared to the conventional algebraic codebook.