ADPCM with a multiquantizer for speech coding
Taniguchi, T.
Unagami, S.
Iseda, K.
Tominaga, S.
Fujitsu Labs. Ltd., Kawasaki;
This paper appears in: Selected Areas in Communications, IEEE Journal on
Publication Date: Feb 1988
Volume: 6,
Issue: 2
On page(s): 410-424
ISSN: 0733-8716
References Cited: 15
CODEN: ISACEM
INSPEC Accession Number: 3150929
Digital Object Identifier: 10.1109/49.616
Current Version Published: 2002-08-06
Abstract
A speech coding algorithm with low complexity and a short
processing delay is introduced. The proposed algorithm is ADPCM
(adaptive digital pulse code modulation) with a multiquantizer
(ADPCM-MQ). The input signal is processed in parallel by multiple ADPCM
coders with different characteristics. Then the optimum ADPCM coder with
minimum error power is dynamically selected for each frame. A 16-kb/s
codec based on this algorithm has been implemented using two
general-purpose digital signal processors (MB8764) with 8.3 ms of total
processing delay. A segmental SNR of 19-21 dB was achieved at 16 kb/s;
with postfiltering the segmental SNR was increased to 23-25 dB. Combined
with the time domain compression scheme, the algorithm can be easily
applied to 8-kb/s coding. It is also extensible to variable-rate coding
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