Ho-Sang Sung

Overview of the EVS codec architecture

The recently standardized 3GPP codec for Enhanced Voice Services (EVS) offers new features and improvements for low-delay real-time communication systems. Based on a novel, switched low-delay speech/audio codec, the EVS codec contains various tools for better compression efficiency and higher quality for clean/noisy speech, mixed content and music, including support for wideband, super-wideband and full-band content. The EVS codec operates in a broad range of bitrates, is highly robust against packet loss and provides an AMR-WB interoperable mode for compatibility with existing systems. This paper gives an overview of the underlying architecture as well as the novel technologies in the EVS codec and presents listening test results showing the performance of the new codec in terms of compression and speech/audio quality.

Packet-loss concealment technology advances in EVS

EVS, the newly standardized 3GPP Codec for Enhanced Voice Services (EVS) was developed for mobile services such as VoLTE, where error resilience is highly essential. The presented paper outlines all aspects of the advances brought during the EVS development on packet loss concealment, by presenting a high level description of all technical features present in the final standardized codec. Coupled with jitter buffer management, the EVS codec provides robustness against late or lost packets. The advantages of the new EVS codec over reference codecs are further discussed based on listening test results.

An efficient LSF quantization using dynamic bit allocation

The current paper is concerned with an effective method to quantize a spectrum envelope of a speech signal without having an inter-frame prediction. In this paper, we proposed a method referred to as dynamic bit allocation-split vector quantization (DBA-SVQ). The main feature of this structure is that it makes use of the ordering property of line spectral frequencies (LSF) and exploits multiple codebooks, normalization and the DBA technique. As a result, we can limit the dynamic range of LSF sub-vectors and allocate different numbers of bits in accordance with the range sizes to maximize the overall efficiency of quantization. The performance is compared with delta line spectral pairs (LSP) VQ, which is used in EVRC-B, demonstrating reduction in spectral distortion (SD).

High-quality audio transform coded excitation using trellis codes

A trellis source code and novel variable-length lossless code are used in transform coded excitation audio coding. Implemented within the extended adaptive multi-rate wideband (AMR-WB+) audio coding framework, the proposed quantization and lossless coding method provide between 0.4 and 0.7 dB increase in signal-to-noise ratio over the E8 lattice VQ and spherical lossless code used in the AMR-WB+ standard.

Low bit rate high-quality MDCT audio coding of the 3GPP EVS standard

This paper presents a low bit-rate MDCT coder, which is adopted as a part of the recently standardized codec for Enhanced Voice Services. To maximize codec performance for NB to SWB input signals for low bit-rates (7.2 to 16.4 kbps), new adaptive bit-allocation and spectrum quantization schemes, which emphasize perceptually important spectrum while efficiently coding full spectrum, was introduced into the low bit-rate MDCT coder. Further, small symbol switched Huffman coding is exploited for reducing the bits consumption for quantizing band energies of the spectrum. Finally, the performance of the coder is illustrated with some listening test results.

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.

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.

Session negotiation and media adaptation of EVS in Voice over LTE

EVS is expected to fill the large gap between the quality and capacity currently available and those expected for 4G, and beyond, mobile communications systems. The large bit-rate and bandwidth ranges, and the complex structures of the new codec make it challenging to deploy and operate. In this paper, we explain how EVS is systematically integrated into the VoLTE ecosystems, outlining session negotiation procedures in which QoS is agreed between two UEs via IMS, and media adaptation procedures in which a controlled compromise is achieved between voice quality and error resilience when the link quality deteriorates or the radio resource runs out.

Super-wideband fine spectrum quantization for low-rate high-quality MDCT coding mode of the 3GPP EVS codec

This article presents a low bit-rate super wideband MDCT coder, which is adopted as a part of the recently standardized codec for Enhanced Voice Services. To maximize codec performance at 13.2 kbps, existing algorithms are reviewed and several new tools are introduced into the low bit-rate MDCT coder to improve the performance of the coder while coding music and mixed content. A subjective listening test demonstrates the advantage of the proposed system for 13.2 kbps when compared to AMR-WB+.