| 摘要 |
<p>A speech coding apparatus includes a spectrum parameter calculation section (200,210), an adaptive codebook section (500), a sound source quantization section (350;355;356;357,366), a discrimination section (370), and a multiplexer section (400). The spectrum parameter calculation section (200,210) receives a speech signal and quantizes a spectrum parameter. The adaptive codebook section (500) obtains a delay and a gain from a past quantized sound source signal using an adaptive codebook, and obtains a residue by predicting a speech signal. The sound source quantization section (350;355;356;357,366) quantizes a sound source signal using the spectrum parameter. The discrimination section (370) discriminates the mode. The sound source quantization section (350;355;356;357,366) has a codebook (351,352) for representing a sound source signal by a combination of non-zero pulses and collectively quantizing amplitudes or polarities of the pulses in a predetermined mode, and searches combinations of code vectors and shift amounts used to shift the positions of the pulses to output a combination of a code vector and shift amount which minimizes distortion relative to input speech. The multiplexer section (400) outputs a combination of outputs from the spectrum parameter calculation section (200,210), the adaptive codebook section (500), and the sound source quantization section (350;355;356;357,366). <IMAGE></p> |
