Browsing by Author "GIBSON, JD"

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FRACTIONAL RATE MULTITREE SPEECH CODING
(1991-06-01)
GIBSON, JD
;
CHANG, WW
;
電信工程研究所
;
Institute of Communications Engineering
We present both forward and backward adaptive speech coders that operate at 9.6, 12, and 16 kb/s using integer and fractional rate trees, weighted squared error distortion measures, the (M,L) tree search algorithm, and incremental path map symbol release. We introduce the concept of multitree source codes and illustrate how the multitree structure allows scalar quantizer-based codes and scalar adaptation rules to be used for fractional rate tree coding. With a frequency weighted distortion measure, the forward and backward adaptive multitree coders produce near toll quality speech at 16 kb/s, while the backward adaptive 9.6 kb/s multitree coder substantially outperforms adaptive predictive coding and has an encoding delay less than 2 ms. Performance results are presented in terms of unweighted and weighted signal-to-noise ratio and segmental signal-to-noise ratio, sound spectrograms, and subjective listening tests.
OBJECTIVE AND SUBJECTIVE OPTIMIZATION OF APC SYSTEM PERFORMANCE
(1990-06-01)
GIBSON, JD
;
CHANG, WW
;
電信工程研究所
;
Institute of Communications Engineering
PATH MAP SYMBOL RELEASE RULES AND THE EXPONENTIAL METRIC TREE
(1990-09-01)
CHANG, WW
;
GIBSON, JD
;
電信工程研究所
;
Institute of Communications Engineering
SMOOTHED DPCM CODES
(1991-09-01)
CHANG, WW
;
GIBSON, JD
;
電信工程研究所
;
Institute of Communications Engineering
Rate distortion theory promises that autoregressive sources can be encoded optimally at small distortions (high rates) by a source coder with infinite encoding delay and zero delay at the decoder. However, for instrumentable systems with finite encoding delay and an unmatched code generator or for operation at low rates, decoding delay may provide a performance increment. The alphabet constrained approach to data compression allows delay at both the encoder and the decoder, and Sethia and Anderson incorporate delay in a tree coder code generator by combining a weighted linear interpolation scheme with DPCM. This system, called interpolative DPCM (IDPCM), was shown to outperform DPCM at rate 1 b/sample for several synthetic source models. In the present work, we use minimum mean squared error (MMSE) fixed-lag smoothing in conjunction with DPCM to develop a code generator employing delayed decoding. This smoothed DPCM (SDPCM) code generator is compared to DPCM and IDPCM code generators at rates 1 and 2 b/sample for tree coding several synthetic sources and to a DPCM code generator at rate 2 b/sample for speech sources. The (M,L) algorithm, which retains only the M best paths to depth L, is used for tree searching, and SDPCM outperforms IDPCM and DPCM at rate 2 b/sample for the synthetic sources with M = 1, 4, 8, and 12, and at rate 1 b/sample with M greater-than-or-equal-to 4. For speech, SDPCM provides a slight improvement in MSE over DPCM codes that is also evident in sound spectrograms and informal subjective listening tests. The models upon which the fixed-lag smoother is based must be chosen appropriately to achieve good SDPCM performance.