We study the application of wavelet packet filterbanks to low bit-rate transparent audio coding, taking the audio coders' delay requirements into account, and propose low-delay coders based on wavelet packet filterbanks. We first develop a method of comparison between filterbanks for perceptual audio coding by estimating the necessary bit-rate for a transparent compression. We use this comparison method in order to select the best filters for our audio compression scheme, from a large set of orthogonal and biorthogonal wavelets. Different wavelet filters may be used at different stages of the tree-structured decomposition with a constraint on the overall delay taken into account. The optimization is carried out with a simulated annealing procedure, proposing two wavelet packet filterbanks, exhibiting average and low delays. They are inserted in a complete audio coder that employs vector quantization and considers psychoacoustic models. The use of the proposed filterbanks leads to the design of a new bit allocation procedure, taking into account the lack of selectivity of the equivalent synthesis filters in a wavelet packet filterbank. The resulting audio scheme is validated through listening tests. The wavelet packet filterbanks are shown to be a promising tool for audio coding, especially for low-delay coding: with average delay, the quality of the wavelet packet filterbanks is as good as with MPEG-1 Layer-2, both with 80 kb/s, and when reducing the delay to 200 samples, 96 kb/s are needed to achieve the same quality.

We present an audio bidimensional encoding scheme. Taking advantage of a new complex filter-bank, and of a regular lattice associated with a new hexagonal projection kernel, this scheme provides each step of the encoder and of the decoder with fast algorithms, which keeps the overall complexity low. Moreover variable or fix length encodings are available without a look-up table. The results show a very good quality at 80 kbit/s for monophonic signals, and a significant improvement with respect to normalized algorithms of a similar complexity.

We address the issue of choosing an optimal wavelet packets decomposition for low-delay audio bit-rate reduction. The decomposition tree and the choice of the wavelet filters for each 2-band cell of the tree are optimized. For a given filterbank we measure the entropy of the symbols resulting from a quantization stage ensuring transparent compression. This measure is the criterion to be minimized under a delay constraint. The optimization is carried out with simulated annealing techniques among a large family of filters. We show that there is no loss choosing the wavelet filters among a family of filters that have been optimized with respect to coding gain for AR(1) models. This provides efficient wavelet packets for low bit-rate low delay audio compression.

Theoretical and practical aspects of the MUSICAM (masking pattern adapted universal subband integrated coding and multiplexing) system are presented. The system is briefly described. It is one of the few codecs able to achieve high audio quality at bit-rates in the range of 64 to 192 kb/s per monophonic channel. It is shown how it has been designed so as to meet the technical requirements of most applications (low delay, low complexity, error robustness, short access units, etc.). Two examples of applications in the field of digital audio broadcasting and multimedia are given.>

The authors address the problem of speech enhancement in a car environment. After describing the method used to characterize the noisy environment, the main effects of the noise on speech processing for speech recognition and transmission are presented. Most of the noise reduction methods proposed in past years have been developed and evaluated in the case of additive white noise. The authors have found that the results obtained by this approach, without modification, are poor. With preprocessing of the noisy signal, improvements are shown, especially for speech recognition.>

The authors propose a low cost 6 kbps multipulse-excited linear prediction coder for 8 kbps transmissions on 12.5 kHz radio channels. The main purpose of this system is to give good intelligibility and good speaker recognition. Against transmission errors, the authors present a correcting strategy and an extrapolation method that hides lost speech frames.>