No abstract

A 64 kbps coder of wideband (15 kHz) monophonic audio signals is described. Its structure is based on the transform coded excitation scheme, adopted to 7 kHz band signals. Significant modifications are proposed, that yield reduction of delay while keeping an almost transparent quality of speech and music, equivalent to that provided by the MPEG1, layer I1 audio standard at the same bit rate. Algorithmic delay has been reduced to 17 ms -approximately 1/3 the delay of the M PEG coder.

In this paper we propose a multiple frequency harmonics model for analysis and synthesis of audio signals. The novelty of this model is that a composite sound can be represented by a few number (two or three) of frequency harmonics, each with time-varying fundamental frequency. We present the model and some key issues in tuning it.Results demonstrate that the model is valid and promising. It is convenient for time-and frequency-scale modifications and is of interest for low bit rate audio coding.

This paper presents a scalable audio format, called "multilayer scalable LPC audio format", that addresses similar functionalities of MPEG-4. The format offers different levels of data rate and audio quality, and answers to the most important requirements of transmission and storage purposes, such as channel error robustness, cell loss robustness, low delay, and playback control. It operates in four modes. The first mode is based on a modified version of the LD-CELP algorithm, in which each 6 samples are represented by one single byte. In order to improve the signal-to-noise ratio (SNR), additional enhancement layers are embedded in the bit stream to allow higher quality at higher bit rates. The resultant bit rates are integer-multiple of 10.67kbps. The other three modes use QMF splitting to two, four and eight subbands. These modes allow efficient representation of wideband audio and speech signals, and offer extension layers of 5.33 and 2.66kbps. A simple and efficient header structure is imbedded in the bitstream to allow the decoding process even in channel error conditions and even when the bitstream has been down-scaled somewhere during the transmission but has not been acknowledged to the decoder. Comparison results are conducted in respect to MPEG and ITU standards.