Markus Multrus

Unified speech and audio coding scheme for high quality at low bitrates

Traditionally, speech coding and audio coding were separate worlds. Based on different technical approaches and different assumptions about the source signal, neither of the two coding schemes could efficiently represent both speech and music at low bitrates. This paper presents a unified speech and audio codec, which efficiently combines techniques from both worlds. This results in a codec that exhibits consistently high quality for speech, music and mixed audio content. The paper gives an overview of the codec architecture and presents results of formal listening tests comparing this new codec with HE-AAC(v2) and AMR-WB+. This new codec forms the basis of the reference model in the ongoing MPEG standardization activity for Unified Speech and Audio Coding.

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.

Efficient context adaptive entropy coding for real-time applications

Context based entropy coding has the potential to provide higher gain over memoryless entropy coding. However serious difficulties arise regarding the practical implementation in real-time applications due to its very high memory requirements. This paper presents an efficient method for designing context adaptive entropy coding while fulfilling low memory requirements. From a study of coding gain scalability as a function of context size, new context design and validation procedures are derived. Further, supervised clustering and mapping optimization are introduced to model efficiently the context. The resulting context modelling associated with an arithmetic coder was successfully implemented in a transform-based audio coder for real-time processing. It shows significant improvement over the entropy coding used in MPEG-4 AAC.

Low-complexity and robust coding mode decision in the EVS coder

Several state-of-the-art switched audio codecs employ the closed-loop mode decision to select the best coding mode at every frame. The closed-loop mode selection is known to have good performance but also high complexity. The new approach we propose in this paper is a low-complexity version of the closed-loop approach, based on similar decisions which compute the coding distortion of each mode and select the one with the lowest distortion. Our approach differs mainly in the way the coding distortions are calculated. We are able to notably reduce the complexity by only estimating the distortions without encoding and decoding the input for each mode. The new approach was implemented in the EVS codec standard and evaluated both objectively and subjectively. Compared to the closed-loop approach, it yields similar performance and lower complexity.

MPEG Unified Speech and Audio Coding – Bridging the Gap

Speech and audio coding schemes originate from different worlds. Speech coding schemes typically assume a source model i.e. the human vocal tract. General audio coding schemes primarily rely on a sinkmodel i.e. the human auditory system. While speech coding schemes work well for the signal class they were designed for at very low rates, they are known to fail for general audio signals even at higher rates. In contrast, general audio coders work well for any content at higher rates, but typically have limited performance especially for speech signals at very low rates. Recently the ISO/MPEG group started a standardization activity to develop a new Unified Speech and Audio Coding scheme. A state of the art AAC based general audio coder, featuring transform coding, parametric bandwidth extension and parametric stereo coding,was extended by source model coding tools. All codec modules were further improved and revised for enhanced performance in particular at very low bitrates. The new unified coding scheme outperforms dedicated speech and general audio coding schemes and bridges the gap between both worlds. This paper describes the new codec in detail and shows how the goal of consistent high quality for all signal types is reached.