Speech encoding method and speech encoding system

• US 6,581,031 B1
• Filed: 11/29/1999
• Issued: 06/17/2003
• Est. Priority Date: 11/27/1998
• Status: Expired due to Term

• Current Assignee: NEC Corporation
• Original Assignee: NEC Corporation
• Inventors: Ozawa, Kazunori, Serizawa, Masahiro, Ito, Hironori
• Primary Examiner(s): ABEBE, DANIEL DEMELASH
• Application Number:
  US09/450,305
• Time in Patent Office: 1,296 Days
• Field of Search: 704/207, 704/208, 704/209, 704/217, 704/219, 704/220, 704/222, 704/223, 704/262, 704/265, 704/501
• US Class Current: 704/222
• CPC Class Codes: G10L 19/08 Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
• CPC Class Codes: G10L 19/09 Long term prediction, i.e. removing periodical redundancies, e.g. by using adaptive codebook or pitch predictor

Abstract

In this speech encoding system, the limiter circuit is input with the delay of adaptive codebook obtained for the previous subframe, and the pitch cycle search range is limited so that the delay of adaptive codebook obtained for the previous subframe is not discontinuous to the delay of adaptive codebook to be obtained for the current subframe, and the pitch cycle search range limited is output to the pitch calculation circuit. The pitch calculation circuit is input with output signal X_w(n) of the perceptual weighting circuit and the pitch cycle search range output from the limiter, calculating the pitch cycle T_op, then outputting at least one pitch cycle T_op to the adaptive codebook circuit. The adaptive codebook circuit is input with the perceptual weighting signal x′_w(n), the past excitation signal v(n) output from the gain quantization circuit, the perceptual weighting impulse response h_w(n) output from the impulse response calculation circuit, and the pitch cycle T_op from the pitch calculation circuit, searching near the pitch cycle, calculating the delay of adaptive codebook. With the above composition, the delay of adaptive codebook obtained for each subframe can be prevented from being discontinuous in the process of time.

Claims

1. A speech encoding method, comprising:

calculating a spectral parameter from a current frame of a speech signal and quantizing said spectral parameter;

calculating a delay and a gain for an adaptive codebook for said current frame of said speech signal using a previously quantized excitation signal from a previous frame of said speech signal;

quantizing an excitation signal of said current frame of said speech signal using said spectral parameter;

quantizing a gain of said excitation signal; and

limiting a search range of an adaptive code vector within a range defined by a pitch position calculated in said previous frame of said speech signal and searching said delay from said current frame of said speech signal.

2. A speech encoding method, comprising:

calculating a spectral parameter from a current frame of a speech signal and quantizing said spectral parameter;

calculating a delay and a gain for an adaptive codebook for said current frame of said speech signal using a previously quantized excitation signal from a previous frame of said speech signal;

quantizing an excitation signal of said current frame of said speech signal using said spectral parameter;

quantizing a gain of said excitation signal;

determining a mode by extracting a characteristic quantity from said speech signal; and

limiting a search range of an adaptive code vector within a range defined by a pitch position calculated in said previous frame of said speech signal and searching said delay from said current frame of said speech signal when said determined mode corresponds to a predetermined mode.

3. A speech encoding system, comprising:

a spectral parameter calculation unit that calculates a spectral parameter from a current frame of a speech signal and quantizes said spectral parameter;

a pitch calculation unit that calculates and outputs a delay from said current frame of said speech signal;

an adaptive codebook unit that calculates a delay and a gain for an adaptive codebook for said current frame of said speech signal using a previously quantized excitation signal from a previous frame of said speech signal, and that outputs said calculated delay and gain;

an excitation quantization unit that quantizes and outputs an excitation signal of said current frame of said speech signal using said spectral parameter;

a gain quantization unit that quantizes and outputs a gain of said excitation signal; and

a limiter unit that limits a search range of an adaptive code vector within a range defined by a pitch position calculated in said previous frame of said speech signal,

wherein said pitch calculation unit outputs a result of searching said delay from said current frame of said speech signal based on the output of said limiter unit.

4. A speech encoding system, comprising:

a spectral parameter calculation unit that calculates a spectral parameter from a current frame of a speech signal and quantizes said spectral parameter;

a pitch calculation unit that calculates and outputs a delay from said current frame of said speech signal;

an adaptive codebook unit that calculates multiple delays and gain for an adaptive codebook for said current frame of said speech signal using a previously quantized excitation signal from a previous frame of said speech signal, and that outputs said calculated delays and gain;

an excitation quantization unit that quantizes an excitation signal of said current frame of said speech signal for each of said multiple delays using said spectral parameter and then outputs an excitation signal with the least signal distortion;

a gain quantization unit that quantizes and outputs the gain of said excitation signal; and

a limiter unit that limits a search range of an adaptive code vector within a range defined by a pitch position calculated in said previous frame of said speech signal,

wherein said pitch calculation unit outputs a result of searching said delay from said current frame of said speech signal based on the output of said limiter unit.

5. A speech encoding system, comprising:

a spectral parameter calculation unit that calculates a spectral parameter from a current frame of a speech signal and quantizes said spectral parameter;

a pitch calculation unit that calculates and outputs a delay from said current frame of said speech signal;

an adaptive codebook unit that calculates a delay and a gain for an adaptive codebook for said current frame of said speech signal using a previously quantized excitation signal from a previous frame of said speech signal, and that outputs said calculated delay and gain;

an excitation quantization unit that quantizes and outputs an excitation signal of said current frame of said speech signal using said spectral parameter;

a mode determination unit that determines a mode by extracting a characteristic quantity from said current frame of said speech signal;

a gain quantization unit that quantizes and outputs the gain of said excitation signal; and

a limiter unit that limits a search range of an adaptive code vector within a range defined by a pitch position calculated in said previous frame of said speech signal when the output of said mode determination unit corresponds to a predetermined mode,

wherein said pitch calculation unit outputs a result of searching said delay from said current frame of said speech signal based on the output of said limiter unit when the output of said mode determination unit corresponds to the predetermined mode.

6. A speech encoding system, comprising:

a spectral parameter calculation unit that calculates a spectral parameter from a current frame of a speech signal and quantizes said spectral parameter;

a pitch calculation unit that calculates and outputs a delay from said current frame of said speech signal;

an adaptive codebook unit that calculates multiple delays and gain for an adaptive codebook for said current frame of said speech signal using a previously quantized excitation signal from a previous frame of said speech signal, and that outputs said calculated delays and gain;

an excitation quantization unit that quantizes an excitation signal of said current frame of said speech signal using said spectral parameter and then outputs an excitation signal with the least signal distortion;

a mode determination unit that determines a mode by extracting a mode by extracting a characteristic quantity from said current frame of said speech signal;

a gain quantization unit that quantizes and outputs the gain of said excitation signal; and

a limiter unit that limits a search range of an adaptive code vector within a range defined by a pitch position calculated in said previous frame of said speech signal when the output of said mode determination unit corresponds to a predetermined mode,

wherein said pitch calculation unit outputs a result of searching said delay from said current frame of said speech signal based on the output of said limiter unit when the output of said mode determination unit corresponds to the predetermined mode.