摘要 |
A stochastic speech coder is described in which real speech is received (4), analysed (13 and 14) to determine coefficients for a vocal tract filter (3), and broken down into frames of samples. For each frame of samples stored in a buffer (5) every stochastic sequence stored in a codebook (1) is scaled (8) and filtered (3) before being compared (9) with the sorted frame. The minimum means (11) square (12) error signal so produced is used to select the optimum sequence so that its index number can be transmitted to a decoder. The identity of the vocal tract filter (3) is also transmitted to the decoder. The analysis circuitry (13, 14) uses vector quantization and this leads to a reduction in the bit rate required to transmit the identity of the vocal tract filter (3), and a simplification of the computation required to select the optimum stochastic sequence. <IMAGE> |