Title :
ADPCM with a multiquantizer for speech coding
Author :
Taniguchi, Tomohiko ; Unagami, Shigeyuki ; Iseda, Kohei ; Tominaga, Syozi
Author_Institution :
Fujitsu Labs. Ltd., Kawasaki, Japan
fDate :
2/1/1988 12:00:00 AM
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
Keywords :
codecs; digital filters; encoding; filtering and prediction theory; pulse-code modulation; speech analysis and processing; vocoders; voice equipment; 16 kbits/s; 23 to 25 dB; 8 kbits/s; ADPCM; adaptive digital pulse code modulation; codec; error power; general-purpose digital signal processors; multiquantizer; postfiltering; processing delay; speech coding algorithm; time domain compression; variable-rate coding; Bit rate; Codecs; Delay; Hardware; Noise level; Performance evaluation; Quantization; Signal to noise ratio; Speech coding; Speech enhancement;
Journal_Title :
Selected Areas in Communications, IEEE Journal on
