Ki-hyun Choo

Bandwidth Extension for China AVS-M standard

We proposed a new frequency domain BandWidth Extension (BWE) technology. In the new technology, FFT based frequency domain gain shaping combined with Linear Prediction Coding (LPC) based spectral envelope shaping is used for generating high frequency signals. To preserve the amount of noise component in the reconstructed band, gain reduction controlled by Spectrum Flatness Measurement (SFM) is employed. Subjective testing results show that the presented technology exhibits a comparable performance compared to 3GPP AMR-WB+ with the same bit-rate in the framework of Audio Video coding of China Standard (AVS) Part 10 - Mobile Speech and Audio Codec. This technology has been formally adopted as the artificial high band coding module in AVS P10.

Two-stage speech/music classifier with decision smoothing and sharpening in the EVS codec

In most internationally recognized standardized multi-mode codecs, signal classification is performed in a single step by either linear discrimination or SNR-based metrics. The speech/music classifier of the EVS codec achieves greater discrimination than these single-step models by combining Gaussian mixture modelling (GMM) with a series of context-based improvement layers. Additionally, unlike traditional GMM classifiers the EVS model adopts a short hangover period, allowing it to track transitions between music and speech. Misclassifications are mitigated by applying a novel decision smoothing and sharpening technique. The results in relatively static environments demonstrate that the new two-stage approach with selective hangover leads to classification accuracies comparable to speech/music classifiers with longer hangovers. They also show that the new approach leads to faster and more accurate switching of coding modes than conventional classifiers for more complex audio environments such as advertisements, jingles and speech superimposed on music.

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.

A novel frequency domain BWE with relaxed synchronization and associated BWE switching

This paper presents a novel frequency domain bandwidth extension (BWE) scheme with relaxed synchronization, optimized for coding inactive and music/mixed content signals. The algorithm achieves high subjective quality at low and medium bitrates and it has a low algorithmic delay. The algorithm is part of the 3GPP Enhanced Voice Services (EVS) codec. In addition to the presented algorithm, the EVS codec employs also a time domain BWE scheme optimized for active speech coding. Consequently a seamless switching between these BWE technologies is required and described in this paper as well.

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+.