Voice coding apparatus, voice decoding apparatus, and voice...

Pulse or digital communications – Pulse position – frequency – or spacing modulation

Reexamination Certificate

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C704S504000, C704S223000

Reexamination Certificate

active

06351490

ABSTRACT:

BACKGROUND OF THE INVENTION
1. Field of the Invention
This invention relates to a voice coding apparatus and a voice decoding apparatus and a voice coding and decoding system, and particularly relates to a voice coding and decoding system capable of establishing a coding rate randomly according to a designated parameter.
This application is based on Patent Application No. Hei 10- 005224 filed in Japan, the content of which is incorporated herein by reference.
2. Background Art
Conventional apparatuses related to this type of voice coding and decoding are used to facilitate to the realization of coding and decoding systems capable of coping with a plurality of applications with varied coding routes by using a single algorithm. For example, a few apparatuses are reported in a paper entitled “A Bit Rate Controllable Voice Coding System MP-CELP” (The Proceedings SSD-5-3 of the Spring Meeting of the Electronic Information Communication Society) (Reference 1) and in the document entitled “Voice Coding Apparatus and Voice Decoding Apparatus” (Japanese Patent Application, First Publication Hei 9-012477 (Reference 2).
These conventional apparatuses are based on CELP (Code Exited Prediction Coding). The CELP system is described in “Code-Excited Linear Prediction: High quality Speech at Very Low Bit Rates (IEEE Proc. ICASSP-85, pp. 937-940, 1985) (Reference 3).
The processing of the conventional apparatus is performed by the steps of: performing linear prediction analysis of the input signals for each frame, calculating the linear prediction factor (LP) representing the spectral envelope characteristics, calculating the drive signal by driving the LP synthetic filter corresponding to the spectral envelope characteristic, and executing coding. The coding of the drive signals is carried out for each subframe which is obtained by further dividing the frame.
Here, the drive signal is composed of a periodic component representing the pitch period of the input signal, a remaining residual component, and their gains. The periodic component representing the pitch period of the input signal is expressed by the adaptive code vector stored in a code book which holds the past drive signals, called an adaptive code book, and the remaining component is expressed as the multi-pulse signals composed of a plurality of pulses.
Furthermore, in the decoding processing of the conventional apparatus, a synthetic voice signal is obtained by inputting the drive signal obtained from the decoded pitch period component and remaining component into the synthetic filter constructed by the decoded LP factors. The positions where each pulse of the multi-pulse signal can exist are restricted in the range shown in a pulse position table, which is called the tracks established in the sub-frames for each pulse. This restriction serves to reduce the number of transmission routes required for coding the pulse position.
A report which describes multi-pulse signals was authored by S. Tamai et al., entitled “Low-Delay CELP with Multi-Pulse VQ and Fast Search for GSM EFR, in ICASSP 96, pp.562-565, May 1996 (Reference 4). The use of multi-pulses makes it easy to switch the coding routing by only changing the number of pulses used for pulse signals. In the conventional apparatuses, it is possible to realize the voice coding and decoding system which is operated at a designated coding route by changing the route setting parameters such as the number of pulses used for the multi-pulses, the length of the frame, or the length of the sub-frame.
Hereinafter, only the length of the sub-frame and the number of pulses will be noted as the route establishing parameter. Thus, the explanation of the apparatus will be executed under the presumption that the other parameters are fixed.
A construction of the conventional voice coding and decoding apparatus will be described hereinafter with reference to
FIGS. 5 and 6
.
FIG. 5
shows a block-diagram showing the structure of a conventional voice coding apparatus. An input terminal
2
is used for inputting a voice signal and delivering the signals to the frame dividing circuit
4
and the sub-frame dividing circuit
10
. The frame dividing circuit
4
cuts the signal supplied from the input terminal
2
into a predetermined length, and delivers it to the LP analysis circuit
6
. The LP analysis circuit
6
obtains the LP factor by analyzing the voice signal supplied from the frame dividing circuit
4
. Furthermore, the LP factor is supplied to an LP factor quantizing circuit
8
, a weighting circuit
12
, and the weighting synthesis circuits
18
,
40
. The detailed explanation of the LP factor analysis is given in the book entitled “Discrete- Time Processing of Speech Signals” (J. R. Deller, MacMillan Pub. 1993)” (Reference 5).
The LP factor quantizing circuit
8
quantizes the LP signals supplied by the LP analysis circuit
6
and the thus obtained codes are delivered to a multiplexer circuit
44
. Furthermore, the quantized LP factors are supplied to the weighting circuit
12
and the weighting synthesis circuits
18
and
40
. A detailed explanation of the quantization is given in “Efficient Vector Quantization of LPC Parameters at
24
Bits IFrame (IEEE Proc. ICASSP-91, pp. 661-664, 1991” (Reference 6).
The input terminal
24
A is used for inputting at the time of starting the coding or for inputting the length of each sub-frame. The sub-frame dividing circuit
10
cuts the voice signal delivered from the input terminal
2
into a length of the sub-frame delivered from the input terminal
24
a
, and supplies it to the weighting circuit
12
. The weighting circuits
12
performs filtering of the voice signal supplied from the LP analysis circuits
6
by the use of an audition weighting filter constructed by the LP factors supplied by the LP analysis circuit
6
. The thus filtered weighted voice signals are delivered to circuit
14
.
The audition weighting filter is described in the reference
3
. The adaptive code book
16
A generates said adaptive code vectors corresponding to the pitch period sequentially delivered from the evaluation circuit
20
, and the adoptive code vectors are delivered to the weighting synthesis circuit
18
. The weighting synthesis circuits
18
execute filtering of adaptive code vectors supplied from the adaptive code book circuit
16
A by use of the audition weighting synthesis filter composed of LP factors supplied by the LP analysis circuit
6
and the quantized LP factors supplied by the LP factor quantization circuit
8
. The thus obtained weighted synthesized signal is delivered to the difference circuit
14
.
The difference circuit
14
calculates the difference between the weighted voice signal supplied by the weighting circuit
12
and the weighted synthesized signal supplied by the weighting synthesizing circuit
18
, and delivers the difference signal to the evaluation circuit
20
. The evaluation circuit
20
delivers a pitch period within a predetermined range to the adaptive code book sequentially, and sequentially calculates the sum of squares of the difference signals supplied from the difference circuit
14
. The code, which corresponds to the pitch period where the sequentially obtained sum of squares becomes a minimum, is delivered to the multiplexer circuit
44
. In addition, the difference signal corresponding to that pitch period is delivered to the difference circuit
22
.
The input terminal
28
is used for inputting the number of pulses at the time of starting the coding or at each sub-frame, and delivers it to a table designing circuit
34
A. The table designing circuit
34
A designs the pulse position table by use of the sub-frame length supplied from the input terminal
24
A and the number of pulses delivered by the input terminal
28
A, and the table is supplied to the table circuit
36
A. The pulse position table is used to make the code correspond with the pulse position. An example will be explained for a case of designing a table when the number of pulses is
5
. When the length of the sub-frame is 40, each pulse position is set

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