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Speech coding and decoding apparatus

Inactive Publication Date: 2000-05-30
KK TOSHIBA
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  • Claims
  • Application Information

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Benefits of technology

With this shortcoming in mind, it is an object of the present invention to provide a speech coding apparatus capable of providing high-quality synthesized sounds even at a low transfer rate.
According to the present invention, in a speech coding apparatus for driving a synthesis filter by an excitation signal to acquire a synthesized sound, the frame of the excitation signal is divided into plural subframes of an equal length or different lengths, a pulse interval is variable subframe by subframe, the excitation signal is formed by a train of excitation pulses with equal intervals in each subframe, the amplitude or the amplitude and phase of the excitation pulse train are determined so as to minimize power of an error signal between an input speech signal and an output signal of the synthesis which is excited by the excitation signal, and the density of the excitation pulse train is determined on the basis of a short-term prediction residual signal or a pitch prediction residual signal to the input speech signal.
According to the present invention, the density or the pulse interval of the excitation pulse train is properly varied in such a way that it becomes dense in those subframes containing important information or many pieces of information and becomes sparse other subframes, thus improving the quality of the synthesized sound.

Problems solved by technology

However, the results of the experiments by this conventional system show that when the transfer rate becomes low, particularly, 10 Kb / s or below, noise in the synthesized sound becomes prominent, deteriorating the quality.
In particular, the quality degradation is noticeable in the experiments with female voices with short pitch.
According to the conventional system, however, even though the power of the prediction residual signal changes within a frame, the synthesis filter is excited by an excitation pulse train always having constant intervals in a frame to acquire a synthesized sound, thus significantly degrading the quality of the synthesized sound.
As described above, since the conventional speech coding system excites the synthesis filter by an excitation pulse train always having constant intervals in a frame, the transfer rate becomes low, 10 Kb / s or lower, for example, the quality of the synthesized sound is deteriorated.

Method used

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first embodiment

FIG. 4 is a block diagram showing a coding apparatus according to the A speech signal s(n) after A / D conversion is input to a frame buffer 102, which accumulates the speech signal s(n) for one frame. Individual elements in FIG. 4 perform the following processes frame by frame.

A prediction parameter calculator 108 receives the speech signal s(n) from the frame buffer 102, and computes a predetermined number, p, of prediction parameters (LPC parameter or reflection coefficient) by an autocorrelation method or covariance method. The acquired prediction parameters are sent to a prediction parameter coder 110, which codes the prediction parameters based on a predetermined number of quantization bits, and outputs the codes to a decoder 112 and a multiplexer 118. The decoder 112 decodes the received codes of the prediction parameters and sends decoded values to a prediction filter 106 and an excitation signal generator 104. The prediction filter 106 receives the speech signal s(n) and an ...

second embodiment

FIG. 8 presents a block diagram of the excitation signal generator 104 using the above excitation pulse calculating algorithm. In FIG. 8, those portions identical to what is shown in FIG. 5 are given the same reference numerals, thus omitting their description.

An impulse response calculator 168 calculates the impulse response hw(n) of the cascade-connection of the synthesis filter and the weighting filter for a predetermined number of samples according to the equation (26) using the quantized value a.sub.i of the prediction parameter input through the input terminal 124 and a predetermined parameter .gamma. of the weighting filter. The acquired hw(n) is sent to a covariance calculator 170 and a correlation calculator 164. The covariance calculator 164 receives the impulse response series hw(n) and calculates covariances .psi..sub.hh (i, j) and .psi..sub.hh (i, j) of hw(n) according to the equations (32) and (31), then sends them to a pulse amplitude calculator 166. A subtracter 171...

fifth embodiment

FIGS. 15 and 16 are block diagrams showing a coding apparatus and a decoding apparatus according to the A frame buffer 11 accumulates one frame of speech signal input to an input terminal 10. Individual elements in FIG. 15 perform the following processes for each frame or each subframe using the frame buffer 11.

A prediction parameter calculator 12 calculates prediction parameters using a known method. When a prediction filter 14 is constituted to have a long-term prediction filter 41 and a short-term prediction filter 42 which are cascade-connected as shown in FIG. 17, the prediction parameter calculator 12 calculates a pitch period, a pitch prediction coefficient, and a linear prediction coefficient (LPC parameter or reflection coefficient) by a known method, such as an autocorrelation method or covariance method. The calculation method is described in, for example, the document 2.

The calculated prediction parameters are sent to a prediction parameter coder 13, which codes the pre...

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Abstract

A speech signal is input to an excitation signal generating section, a prediction filter and a prediction parameter calculator. The prediction parameter calculator calculates a predetermined number of prediction parameters (LPC parameter or reflection coefficient) by an autocorrelation method or covariance method, and supplies the acquired prediction parameters to a prediction parameter coder. The codes of the prediction parameters are sent to a decoder and a multiplexer. The decoder sends decoded values of the codes of the prediction parameters to the prediction filter and the excitation signal generating section. The prediction filter calculates a prediction residual signal, which is the difference between the input speech signal and the decoded prediction parameter, and sends it to the excitation signal generating section. The excitation signal generating section calculates the pulse interval and amplitude for each of a predetermined number of subframes based on the input speech signal, the prediction residual signal and the quantized value of the prediction parameter, and sends them to the multiplexer. The multiplexer combines these codes and the codes of the prediction parameters, and send the results as an output signal of a coding apparatus to a transmission path or the like.

Description

TECHNICAL FIELDThe present invention relates to a speech coding apparatus which compresses a speech signal with a high efficiency and decodes the signal. More particularly, this invention relates to a speech coding apparatus based on a train of adaptive density excitation pulses and whose transfer bit rate can be set low, e.g., to 10 Kb / s or lower.BACKGROUND ARTTodays, coding technology for transferring a speech signal at a low bit rate of 10 Kb / s or lower has been extensively studied. As a practical method is known using a system in which an excitation signal of a speech synthesis filter is represented by a train of pulses aligned at predetermined intervals and the excitation signal is used for coding the speech signal. The details of this method are explained in the paper titled "Regular-Pulse Excitation--A Novel Approach to Effective and Efficient Multipulse Coding of Speech," written by Peter Kroon et al. in the IEEE Report, October 1986, Vol. ASSP-34, pp. 1054-1063 (Document 1)...

Claims

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Application Information

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IPC IPC(8): G10L19/04G10L19/10G10L19/113
CPCG10L19/113
Inventor AKAMINE, MASAMIMISEKI, KIMIO
Owner KK TOSHIBA
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