Abstract
This paper presents a transform coding algorithm designed for audio coding at a bit rate of 64 kbit/s. It enables the transmission of a high quality stereo sound through the 2B channels of isdn. Although a complete system including framing, synchronization and error correction has been developed, only the bit rate compression algorithm is described here. A detailed analysis of the signal processing techniques such as the time/frequency transformation, the preecho reduction by adaptive filtering, the fast algorithm computations…, is provided. The use of psychoacoustical properties is also precisely reported. Finally, some subjective evaluation results and one real time implementation of the coder using the att dsp52c digital signal processor are presented.
Résumé
Cet article présente un algorithme de codage par transformation pour la transmission de signaux sonores de haute qualité à 64 kbit/s. Il permet le transport d’un son stéréo de haute qualité au travers de l’accès de base du rnis. Quoiqu’un système de codage complet incluant la synchronisation, la protection contre les erreurs…, ait été développé, seul l’algorithme de compression de débit est décrit ici. Une analyse détaillée des techniques de traitement du signal employées : transformation temps/fréquence, réduction des prééchos par filtrage adaptatif, algorithmes rapides,… est fournie. La prise en compte des propriétés psychoacoustiques dans l’algorithme de codage est également décrite en détail. Enfin, les performances de ce codeur en termes de qualité subjective sont évoquées ainsi que son implantation sur processeur de signal.
Similar content being viewed by others
References
Brandenburg (K.), Herre (H.), Johnston (J.), Mahieux (Y.), Schroeder (E.). ASPEC : Adaptive perceptual entropy coding of high quality music signals.Proceedings of the 90th AES convention (1991), Paris, Preprint 3011, pp. 1–11.
Brandenburg (K.). High quality sound coding at 2.5 bit/sample.The 84th AES convention (1988), Paris, Preprint 2582, pp. 1–14.
Stoll (G.), Dehery (Y.). High quality bit rate reduction system family for different applications.Proceedings of ICC (1990), pp. 937–941.
Musmann (H.). The ISO coding standard.Proceedings of Globecorn 90, San Diego, pp. 511–517.
Portnoff (M.). Time frequency representation of digital signals.IEEE Trans. ASSP (1980), 28, pp. 55–69.
Jayant (N.), Noll (P.). Digital coding of waveforms.Prentice Hall (1984).
Zelinski (R.), Noll (P.). Adaptive transform coding of speech.IEEE Trans. ASSP (1977), 25, pp. 299–309.
Lookabaugh (T.), Perkins (M). Analysis/Synthesis systems in the presence of quantization.Proceedings of ICASSP, Glasgow (1989), pp. 1341–1344.
Princen (J.), Johnson (A.), Bradley (A.). Adaptive transform coding incorporating time domain aliasing cancellation.Speech Communication (1987), 6, pp. 299–308.
Malvar (H.). Lapped transforms for efficient transform/subband coding.IEEE Trans. ASSP (1990), 38, no 6, pp. 969–978.
Duhamel (P.), Mahieux (Y.), Petit (J. P.). A fast algorithm for the implementation of filter banks based on time domain aliasing cancellation.Proceedings of ICASSP (1991), Toronto, pp. 2209–2112.
Duhamel (P.). Un algorithme de transformation de Fourier rapide à double base.Ann. des Télécommunic (1985), 40, no 9, pp. 481–494.
Schroeder (M.), Atal (B.), Hall (J.). Optimizing digital speech coders by exploiting masking properties of the human ear.Journal of Acoustical Society of America (1979), 66, no 6, pp. 1647–1652.
Zwicker (E.), Feldtkeller (E.). Psychoacoustique.CTST Masson (1981).
Botte (M.), Canevet (G.), Demany (L.), Sorin (C). Psychoacoustique et perception auditive.Editions INSERM (1989).
Brandenburg (K.), Johnston (J.). Second generation perceptual audio coding : The hybrid coder.Proceedings of AES 88th convention (1990), Preprint 2937, pp. 1–12.
Mahieux (Y), Petit (J. P.). Transform coding of audio signal at 64 kbit/s.Proceedings of Globecom 90, San Diego, pp. 518–522.
Perkins (M), Lookabaugh (T.). A psychophysically justified bit allocation algorithm.Proceedings of ICASSP (1989), Glasgow, pp. 1815–1818.
Bozic (S.). Digital and Kaiman filtering.Edward Arnold (1979).
Charbonnier (A.), Petit (J. P.). Subband adpcm coding for high quality audio signals.Proceedings of ICASSP (1988), New York, pp. 2540–2543.
ISO/IEC JTC1/SC2/WG11. MPEG Audio subjective assessments. Test reports.MPEG Document 90/196 (1990).
Author information
Authors and Affiliations
Rights and permissions
About this article
Cite this article
Mahieux, Y. High quality audio transform coding at 64 kbit/s. Ann. Télécommun. 47, 95–106 (1992). https://doi.org/10.1007/BF02999682
Received:
Accepted:
Issue Date:
DOI: https://doi.org/10.1007/BF02999682
Key words
- Speech coding
- Sound signal
- Sound quality
- Passband compression
- Signal processing
- Psychoacoustic
- Cosine transformation
- Discrete transformation