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Highly efficient low-rate audio coding methods are required for new compelling and commercially interesting applications of streaming, messaging and broadcasting services using audio media in 3rd generation mobile communication systems. After an audio codec selection phase 3GPP has standardized the extended AMR-WB (AMR-WB+) codec that provides unique performance at very low bit rates from below 10 kbps up to 24 kbps. This paper discusses the requirements imposed by mobile audio services and gives a technology overview of AMR-WB+ as a codec matching these requirements while providing outstanding audio quality.

A b s t r a c tThe application of Algebraic Code Excited Linear Prediction (ACELP) coding to wideband speech is presented. In wideband coding, a very large excitation codebook is required in order to obtain high quality speech. Ordinary CELP algorithms fail to accomodate such large codebooks due to the excessive coding complexity. In ACELP, however, an algebraic codebook with 20 bit address can be used, without any storage requirements, and more importantly, with a very efficient search procedure which allows for real-time implementation. A focused search strategy is used in which a very small portion of the codebook is searched, yet with performance very close t o that of full search. High quality speech was obtained a t bit rates below 13 kb/s.

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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