Speech data compression through sparse coding of innovations

@article{Ramabadran1994SpeechDC,
  title={Speech data compression through sparse coding of innovations},
  author={Tenkasi V. Ramabadran and Deepen Sinha},
  journal={IEEE Trans. Speech and Audio Processing},
  year={1994},
  volume={2},
  pages={274-284}
}
A new scheme for coding speech at low bit rates (4.8-16 kbls) but still maintaining high quality is described. Speech is regarded as a piecewise-stationary random signal and its synthesis is accomplished by means of a Kalman estimator at the decoder. The Kalman estimator requires for its operation a signal model and a sequence of measurements of the states of the model. A two-stage, time-varying, all-pole filter excited by white noise is used as the speech signal model. Linear combinations of… CONTINUE READING

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