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.

ITU-T G.729.1 is a scalable coder recently standardized in ITU-T for wideband telephony and voice over IP (VoIP) applications. Composed of three stages, this codec provides a scalable bitstream between 8 and 32 kbit/s both in narrowband and wideband. This paper describes the first stage which is a narrowband embedded CELP coder at 8 and 12 kbit/s. The 8 kbit/s layer ensures interoperability with ITU-T G.729 standard with a reduced complexity, and with a quality better than G.729 Annex A. At 12 kbit/s, G.729.1 reaches the quality level of the 11.8 kbit/s G.729 Annex E in spite of the embedded structure. The modifications brought to the original G.729 scheme to achieve this performance are explained and formal test results provided.

This paper presents a hybrid audio coding algorithm integrating an LP-based coding technique and a more general transform coding technique. ACELP is used in LP-based coding mode, whereas algebraic TCX is used in transform coding mode. The algorithm extends previously published work on ACELP/TCX coding in several ways. The frame length is increased to 80 ms, adaptive multi-length sub-frames are used with overlapping windowing, an extended multi-rate algebraic VQ is applied to the TCX spectrum to avoid quantizer saturation, and noise shaping is improved. Results show that the proposed hybrid coder has consistently high performance for both speech and music signals.

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 a 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.

The description and design of the source-controlled variable-rate multimode wideband (VMR-WB) codec recently selected by the 3/sup rd/ Generation Partnership Project 2 (3GPP2) for the cdma2000/spl reg/ system in Rate-Set II are presented. The paper gives an overview of the codec and the methodologies that enable high quality wideband coding at average data rates ranging from TIA/EIA/IS-733 ADR (average data rate) to that of TIA/EIA/IS-127. The codec has three modes of operation at different average data rates and a fourth mode that is interoperable with 3GPP/AMR-WB (ITU-T G.722.2). Despite the interoperability constraint, the codec is capable of meeting the aggressive performance requirements through the use of novel techniques such as noise suppression, efficient signal classification, new coding types optimized for stable voiced and unvoiced frames, novel post-processing technique for periodicity enhancement in the lower frequency band, and improved frame erasure concealment mechanisms.

We present a new method, called Voronoi extension, for the design of low-complexity multi-rate lattice vector quantization (VQ). With this technique, lattice codebooks of arbitrarily large bit rates can be generated algorithmically and the problem of lattice codebook overload can be bypassed. We describe a practical multi-rate quantization system based on Voronoi extension and derived from the lattice RE/sub 8/. This system is applied to the TCX coding model using pitch prediction, so as to extend AMR-WB speech coding at high bit rates (in particular 32 kbit/s).

This paper describes the adaptive multirate wideband (AMR-WB) speech codec selected by the Third Generation Partnership Project (3GPP) for GSM and the third generation mobile communication WCDMA system for providing wideband speech services. The AMR-WB speech codec algorithm was selected in December 2000 and the corresponding specifications were approved in March 2001. The AMR-WB codec was also selected by the International Telecommunication Union-Telecommunication Sector (ITU-T) in July 2001 in the standardization activity for wideband speech coding around 16 kb/s and was approved in January 2002 as Recommendation G.722.2. The adoption of AMR-WB by ITU-T is of significant importance since for the first time the same codec is adopted for wireless as well as wireline services. AMR-WB uses an extended audio bandwidth from 50 Hz to 7 kHz and gives superior speech quality and voice naturalness compared to existing second- and third-generation mobile communication systems. The wideband speech service provided by the AMR-WB codec will give mobile communication speech quality that also substantially exceeds (narrowband) wireline quality. The paper details AMR-WB standardization history, algorithmic description including novel techniques for efficient ACELP wideband speech coding and subjective quality performance of the codec.

We present in this paper new methods for achieving high- quality wideband speech at low rates using the ACELP algorithm. Several innovations are introduced to optimize the quality and minimize the complexity of the coder. A multi-rate wideband speech encoding algorithm based on these techniques was recently selected by 3GPP as the standard for AMR-WB, and is currently one of the candidates for the ITU- T wideband speech coder standard at around 16 kbit/sec. This standard was jointly developed by VoiceAge and Nokia.

The Third Generation Partnership Project (3GPP) and European Telecommunications Standards Institute (ETSI) have carried out development and standardisation of a wideband speech codec for GSM and the third generation mobile communication WCDMA system since 1999. The Adaptive Multi-Rate Wideband (AMR-WB) codec algorithm was selected in December 2000, and the corresponding specifications were approved in March 2001. The AMR-WB codec was jointly developed by Nokia and VoiceAge. AMRWB extends the audio bandwidth from 3.4 kHz to 7 kHz and gives superior speech quality and voice naturalness compared to existing 2nd and 3rd generation mobile communication systems. The wideband speech service provided by the AMRWB codec will give mobile communication speech quality that even exceeds (narrowband) wireline quality.

This paper describes the recently adopted ITU-T Recommendation G.729 Annex A (G.729A) for encoding speech signals at 8 kbit/s with low complexity. G.729A has been selected as the standard speech coding algorithm for multimedia digital simultaneous voice and data (DSVD). G.729A is bitstream interoperable with G.729; i.e., speech coded with G.729A can be decoded with G.729, and vice versa. As G.729, it uses the conjugate structure algebraic code excited linear prediction (CS-ACELP) algorithm with 10 ms frames. However, several algorithmic changes have been introduced into G.729 which resulted in 50% drop in its complexity, enabling a DSP implementation with a complexity of about 10-12 MIPS. This paper describes the algorithmic changes which have been introduced in order to achieve the low complexity goal while meeting the terms of reference. Subjective tests have been performed by ITU-T in both the selection phase and the characterization phase and the results showed that the performance of G.729A is equivalent to both G.729 and G.726 at 32 kbit/s in most operating conditions; however, it is slightly worse in case of three tandems and in the presence of background noise. A breakdown of the complexities of both G.729 and G.729A is given at the end of the paper.