Subband/Transform coding using filter bank designs based on time domain aliasing cancellation

@article{Princen1987SubbandTransformCU,
  title={Subband/Transform coding using filter bank designs based on time domain aliasing cancellation},
  author={J. Princen and A. Johnson and A. Bradley},
  journal={ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing},
  year={1987},
  volume={12},
  pages={2161-2164}
}
A new, oddly stacked, critically sampled, single side-band (SSB) [7] analysis/synthesis system based on Time Domain Aliasing Cancellation (TDAC) [1],[2] is described in this paper. The specifications for the analysis and synthesis filter responses are developed and a number of designs which satisfy the reconstruction requirements are described. The application of TDAC systems to Subband/Transform coding is also discussed and the objective performance of a 32 band coder using several different… Expand
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References

SHOWING 1-10 OF 10 REFERENCES
Analysis/Synthesis filter bank design based on time domain aliasing cancellation
TLDR
A single-sideband analysis/synthesis system is proposed which provides perfect reconstruction of a signal from a set of critically sampled analysis signals and allows overlap between adjacent time windows, implying that time domain aliasing is introduced in the analysis; however, thisAliasing is cancelled in the synthesis process, and the system can provide perfect reconstruction. Expand
Polyphase quadrature filters-A new subband coding technique
TLDR
A new method of implementing filter banks for subband coding of speech by combining the quadrature filter characteristic with the polyphase network implementation of filter banks, which requires 35% fewer computations than existing designs. Expand
Frequency domain coding of speech
Frequency domain techniques for speech coding have recently received considerable attention. The basic concept of these methods is to divide the speech into frequency components by a filter bankExpand
Design and applications of uniform digital bandpass filter banks
A digital bandpass filter-bank for demodulating a wideband frequency multiplexed signal into a specified number of uniform narrow band channel outputs, and conversely for modulating a set of channelExpand
Quadrature mirror filter design for an arbitrary number of equal bandwidth channels
  • P. Chu
  • Mathematics, Computer Science
  • IEEE Trans. Acoust. Speech Signal Process.
  • 1985
TLDR
A key feature of this filter structure is that the number of multiplies, adds, and stored coefficients required for implementation is significantly less than those needed for the conventional QMF structure, given the same number of channels. Expand
Implementation of the digital phase vocoder using the fast Fourier transform
TLDR
This paper discusses a digital formulation of the phase vocoder, an analysis-synthesis system providing a parametric representation of a speech waveform by its short-time Fourier transform, designed to be an identity system in the absence of any parameter modifications. Expand
Adaptive quantization with a one-word memory
We discuss a quantizer which, for every new input sample, adapts its step-size by a factor depending only on the knowledge of which quantizer slot was occupied by the previous signal sample.1Expand
A weighted overlap-add method of short-time Fourier analysis/Synthesis
TLDR
A new structure and a simplified interpretation of short-time Fourier synthesis using synthesis windows is presented and it is shown how this structure can be used for analysis/synthesis applications which require different analysis and synthesis rates, such as time compression or expansion. Expand
"J."
however (for it was the literal soul of the life of the Redeemer, John xv. io), is the peculiar token of fellowship with the Redeemer. That love to God (what is meant here is not God’s love to men)Expand
N . S . Jayant , " Adaptive Quantization with a One Word Memory " , BSTJ . Vol . 52 , No . 7
  • 1978