A 6Kbps to 85Kbps scalable audio coder

  • Authors:
  • T. S. Verma;T. H. Y. Meng

  • Affiliations:
  • Lab. of Acoustics & Audio Signal Processing, Helsinki Univ. of Technol., Espoo, Finland;-

  • Venue:
  • ICASSP '00 Proceedings of the Acoustics, Speech, and Signal Processing, 2000. on IEEE International Conference - Volume 02
  • Year:
  • 2000

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Abstract

Scalable audio coding is important in network environments, such as the Internet, where bandwidth is not guaranteed, packet loss is common, and client connection data rates are heterogeneous. Signal models provide a general framework for attacking a wide range of challenges in the unicast delivery of real-time audio over packet switched networks. The specific signal model in this work generates a parametric representation for general wide-band audio signals. The model consists of three complementary components: sines, transients, and noise. Because the human hearing system ultimately judges the validity of a model for audio signals, psychoacoustic principles are explicitly considered in the three part model. Once analyzed, the parameters are quantized, compressed and packed into a single 85Kbps bit-stream. From this bit-stream, bit-streams at several bit-rates between 6Kbps and 85Kbps may be readily extracted. The audio coder offers a wide range of scalability while the audio quality of the coding scheme gracefully degrades from perceptually lossless to low-quality.