Name
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 |
Year | DOI | Venue |
|---|---|---|
1988 | 10.1109/49.616 | IEEE Journal on Selected Areas in Communications |
Keywords | Field | DocType |
Speech coding,Bit rate,Delay,Codecs,Speech enhancement,Quantization,Performance evaluation,Hardware,Signal to noise ratio,Noise level | Pulse-code modulation,Speech coding,Data transmission,Computer science,Speech recognition,Adaptive filter,Quantization (signal processing),Codec,Adaptive coding,Processing delay | Journal |
Volume | Issue | ISSN |
6 | 2 | 0733-8716 |
Citations | PageRank | References |
1 | 0.63 | 1 |
Authors | ||
4 | ||
Name | Order | Citations | PageRank |
|---|---|---|---|
| T. Taniguchi | 1 | 6 | 1.92 |
| S. Unagami | 2 | 1 | 0.97 |
| K. Iseda | 3 | 1 | 0.63 |
| S. Tominaga | 4 | 1 | 0.63 |