Julien Robilliard

Efficient transform coding of two-channel audio signals by means of complex-valued stereo prediction

Traditional MDCT-based perceptual audio coding schemes employ mid/side and intensity stereo techniques to allow efficient joint coding of the two channels of a stereophonic signal. These techniques, however, provide only little coding gain for critical stereo signals characterized by spectral components with a distinct level or phase difference between the channels. To overcome this deficiency, we propose an extension to the mid/side coding paradigm that utilizes complex-valued inter-channel linear prediction in the MDCT spectral domain. The required imaginary spectrum (MDST) is calculated in a computationally efficient manner without additional algorithmic delay. A formal listening test conducted in the course of the ISO/MPEG standardization of the unified speech and audio codec USAC illustrates that the proposed stereo prediction approach provides significant improvements in coding efficiency and shows that at 96 kb/s, excellent quality can be obtained even for critical signals.

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.