Source: https://patents.google.com/patent/US8688439B2/en
Timestamp: 2018-07-18 15:00:56
Document Index: 268757086

Matched Legal Cases: ['§371', '§120', '§119', 'Application No. 9', 'Application No. 2', 'Application No. 09014422']

US8688439B2 - Method for speech coding, method for speech decoding and their apparatuses - Google Patents
US8688439B2
US8688439B2 US13792508 US201313792508A US8688439B2 US 8688439 B2 US8688439 B2 US 8688439B2 US 13792508 US13792508 US 13792508 US 201313792508 A US201313792508 A US 201313792508A US 8688439 B2 US8688439 B2 US 8688439B2
US13792508
US20130204615A1 (en )
A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. Ina speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation code books are used based on an evaluation result.
This application is a Divisional of application Ser. No. 12/332,601, filed on Dec. 11, 2008, which is a Divisional of application Ser. No. 11/976,841, filed on Oct. 29, 2007, which is a Continuation of application Ser. No. 11/653,288 (now issued), filed on Jan. 16, 2007, which is a divisional of application Ser. No. 11/188,624 (now issued), filed on Jul. 26, 2005, which is a divisional of application Ser. No. 09/530,719 filed May 4, 2000 (now issued), which is the national phase under 35 U.S.C. §371 of PCT International Application No. PCT/JP98/05513 having an international filing date of Dec. 7, 1998 and designating the United States of America and for which priority is claimed under 35 U.S.C. §120, said PCT International Application claiming priority under 35 U.S.C. §119(a) of Application No. 9-354754 filed in Japan on Dec. 24, 1997, the entire contents of all above-mentioned applications being incorporated herein by reference.
In the improved speech coding and decoding method illustrated in FIG. 7 according to the related art, the plurality of excitation codebooks is switched based on the state of the input speech for producing a coded speech. Therefore, it is possible to use an excitation codebook including noise time series vectors in an unvoiced noise period of the input speech and an excitation code book including non-noise time series vectors in a voiced period other than the unvoiced noise period, for example. Hence, even if a noise speech is coded and synthesized, an unnatural sound, e.g., “Jiri-Jiri,” is not produced. However, since the excitation codebook used in coding is also used in decoding, it becomes necessary to code and transmit data which excitation codebook was used. It becomes an obstacle for lowering bit rates.
In a speech coding method according to another invention, a first excitation codebook storing a noise time series vector and a second excitation codebook storing a non-noise time series vector are provided. A time series vector is generated by adding the time series vector in the first excitation code book and the time series vector in the second excitation codebook by weighting based on an evaluation result of a noise level of a speech.
In a speech decoding method according to another invention, a first excitation codebook storing a noise time series vector and a second excitation codebook storing a non-noise time series vector are provided. A time series vector is generated by adding the time series vector in the first excitation code book and the time series vector in the second excitation codebook by weighting based on an evaluation result of a noise level of a speech.
A speech coding apparatus according to another invention includes a spectrum information encoder for coding spectrum information of an input speech and outputting a coded spectrum information as an element of a coding result, a noise level evaluator for evaluating a noise level of a speech in a concerning coding period by using a code or coding result of at least one of the spectrum information and power information, which is obtained from the coded spectrum information provided by the spectrum information encoder, and outputting an evaluation result, a first excitation codebook storing a plurality of non-noise time series vectors, a second excitation code book storing a plurality of noise time series vectors, an excitation codebook switch for switching the first excitation codebook and the second excitation codebook based on the evaluation result by the noise level evaluator, a weighting-adder for weighting the time series vectors from the first excitation code book and second excitation codebook depending on respective gains of the time series vectors and adding a synthesis filter for producing a coded speech based on an excitation signal, which are weighted time series vectors, and the coded spectrum information provided by the spectrum information encoder, and a distance calculator for calculating a distance between the coded speech and the input speech, searching an excitation code and gain for minimizing the distance, and outputting a result as an excitation code, and a gain code as a coding result.
FIG. 1 illustrates a whole configuration of a speech coding method and speech decoding method in embodiment 1 according to this invention. In FIG. 1, an encoder 1, a decoder 2, a multiplexer 3, and a divider 4 are illustrated. The encoder 1 includes a linear prediction parameter analyzer 5, linear prediction parameter encoder 6, synthesis filter 7, adaptive codebook 8, gain encoder 10, distance calculator 11, first excitation codebook 19, second excitation codebook 20, noise level evaluator 24, excitation code book switch 25, and weighting-adder 38. The decoder 2 includes a linear prediction parameter decoder 12, synthesis filter 13, adaptive codebook 14, first excitation code book 22, second excitation codebook 23, noise level evaluator 26, excitation code book switch 27, gain decoder 16, and weighting-adder 39. In FIG. 1, the linear prediction parameter analyzer 5 is a spectrum information analyzer for analyzing an input speech S1 and extracting a linear prediction parameter, which is spectrum information of the speech. The linear prediction parameter encoder 6 is a spectrum information encoder for coding the linear prediction parameter, which is the spectrum information and setting a coded linear prediction parameter as a coefficient for the synthesis filter 7. The first excitation codebooks 19 and 22 store pluralities of non-noise time series vectors, and the second excitation code books 20 and 23 store pluralities of noise time series vectors. The noise level evaluators 24 and 26 evaluate a noise level, and the excitation codebook switches 25 and 27 switch the excitation code books based on the noise level.
An old excitation signal is stored in the adaptive codebook 8, and a time series vector corresponding to an adaptive code inputted by the distance calculator 11, which is generated by repeating an old excitation signal periodically, is outputted. The noise level evaluator 24 evaluates a noise level in a concerning coding period based on the coded linear prediction parameter inputted by the linear prediction parameter encoder 6 and the adaptive code, e.g., a spectrum gradient, short-term prediction gain, and pitch fluctuation as shown in FIG. 2, and outputs an evaluation result to the excitation codebook switch 25. The excitation codebook switch 25 switches excitation code books for coding based on the evaluation result of the noise level. For example, if the noise level is low, the first excitation code book 19 is used, and if the noise level is high, the second excitation code book 20 is used.
A plurality of non-noise time-series vectors, e.g., a plurality of time series vectors generated by training for reducing a distortion between a speech for training and its coded speech, is stored in the first excitation code book 22. A plurality of noise time series vectors, e.g., a plurality of vectors generated from random noises, is stored in the second excitation codebook 23. Each of the first and second excitation code books outputs a time series vector respectively corresponding to an excitation code. The time series vectors from the adaptive code book 14 and one of first excitation codebook 22 or second excitation codebook 23 are weighted by using respective gains, decoded from gain codes by the gain decoder 16, and added by the weighting-adder 39. An addition result is provided to the synthesis filter 13 as an excitation signal, and an output speech S3 is produced. These operations are characteristic operations in the speech decoding method in embodiment 1.
The excitation codebook 28 stores a plurality of time series vectors generated from random noises, for example, and outputs a time series vector corresponding to an excitation code inputted by the distance calculator 11. If the noise level is low in the evaluation result of the noise, the sampler 29 outputs a time series vector, in which an amplitude of a sample with an amplitude below a determined value in the time series vectors, inputted from the excitation code book 28, is set to zero, for example. If the noise level is high, the sampler 29 outputs the time series vector inputted from the excitation codebook 28 without modification. Each of the time series vectors from the adaptive codebook 8 and the sampler 29 is weighted by using a respective gain provided by the gain encoder 10 and added by the weighting-adder 38. An addition result is provided to the synthesis filter 7 as excitation signals, and a coded speech is produced. The distance calculator 11 calculates a distance between the coded speech and the input speech S1, and searches an adaptive code, excitation code, and gain for minimizing the distance. When coding is over, the linear prediction parameter code and the adaptive code, excitation code, and gain code for minimizing a distortion between the input speech and the coded speech are outputted as a coding result S2. These are characteristic operations in the speech coding method in embodiment 3.
In embodiment 3, the excitation codebook storing noise time series vectors is provided, and an excitation with a low noise level can be generated by sampling excitation signal samples based on an evaluation result of the noise level of the speech. Hence, a high quality speech can be reproduced with a small data amount. Further, since it is not necessary to provide a plurality of excitation codebooks, a memory amount for storing the excitation code book can be reduced.
The first excitation code book 32 stores a plurality of noise time series vectors generated from random noises, for example, and outputs a time series vector corresponding to an excitation code. The second excitation code book 33 stores a plurality of time series vectors generated by training for reducing a distortion between a speech for training and its coded speech, and outputs a time series vector corresponding to an excitation code inputted by the distance calculator 11. The weight determiner 34 determines a weight provided to the time series vector from the first excitation codebook 32 and the time series vector from the second excitation codebook 33 based on the evaluation result of the noise level inputted from the noise level evaluator 24, as illustrated in FIG. 5, for example. Each of the time series vectors from the first excitation codebook 32 and the second excitation codebook 33 is weighted by using the weight provided by the weight determiner 34, and added. The time series vector outputted from the adaptive codebook 8 and the time series vector, which is generated by being weighted and added, are weighted by using respective gains provided by the gain encoder 10, and added by the weighting-adder 38. Then, an addition result is provided to the synthesis filter 7 as excitation signals, and a coded speech is produced. The distance calculator 11 calculates a distance between the coded speech and the input speech S1, and searches an adaptive code, excitation code, and gain for minimizing the distance. When coding is over, the linear prediction parameter code, adaptive code, excitation code, and gain code for minimizing a distortion between the input speech and the coded speech, are outputted as a coding result.
The first excitation codebook 35 and the second excitation codebook 36 output time series vectors corresponding to excitation codes. The weight determiner 37 weights are based on the noise level evaluation result inputted from the noise level evaluator 26 in a same method with the weight determiner 34 in the encoder 1. Each of the time series vectors from the first excitation code book 35 and the second excitation codebook 36 is weighted by using a respective weight provided by the weight determiner 37, and added. The time series vector outputted from the adaptive codebook 14 and the time series vector, which is generated by being weighted and added, are weighted by using respective gains decoded from the gain codes by the gain decoder 16, and added by the weighting-adder 39. Then, an addition result is provided to the synthesis filter 13 as an excitation signal, and an output speech S3 is produced.
1. A speech decoding method for an apparatus having a decoder, the method comprising:
receiving, by the decoder, a coded speech signal including a gain code;
decoding, by the decoder, a gain from the gain code;
modifying the time series vector based on the decoded gain such that the number of samples with zero amplitude-value is changed; and
synthesizing a speech signal based on the modified time series vector.
obtaining an adaptive code vector from an adaptive codebook based on an adaptive code associated with the coded speech signal.
weighting the adaptive code vector and the modified time series vector; and
adding together the weighted adaptive code vector and the weighted time series vector.
decoding a linear prediction parameter from a linear prediction parameter code associated with the coded speech signal; and
synthesizing the speech signal using the linear prediction parameter and the added weighted adaptive code vector and weighted time series vector.
5. The method of claim 4, wherein the decoded linear prediction parameter corresponds to coefficients of a synthesis filter.
6. The method of claim 2, wherein the gain is decoded in a decoding period corresponding to the coded speech signal.
7. The method of claim 2, wherein the adaptive codebook is based on a past excitation.
weighting an adaptive code vector and the modified time series vector; and
10. The method of claim 1, wherein the time series vector is modified based on a noise level.
11. A speech decoding apparatus having a decoder configured to:
receive, by the decoder, a coded speech signal including a gain code;
decode, by the decoder, a gain from the gain code;
modify the time series vector based on the decoded gain such that the number of samples with zero amplitude-value is changed: and
synthesize a speech signal based on the modified time series vector.
obtain an adaptive code vector from an adaptive codebook based on an adaptive code associated with the coded speech signal.
weight the adaptive code vector and the modified time series vector; and
add together the weighted adaptive code vector and the weighted time series vector.
decode a linear prediction parameter from a linear prediction parameter code associated with the coded speech signal; and
synthesize the speech signal using the linear prediction parameter and the added weighted adaptive code vector and weighted time series vector.
15. The apparatus of claim 14, wherein the decoded linear prediction parameter corresponds to coefficients of a synthesis filter.
16. The apparatus of claim 12, wherein the gain is decoded in a decoding period corresponding to the coded speech signal.
17. The apparatus of claim 12, wherein the adaptive codebook is based on a past excitation.
18. The apparatus of claim 11, further configured to:
weight an adaptive code vector and the modified time series vector; and
20. The apparatus of claim 11, wherein the time series vector is modified based on a noise level.
US13792508 1997-12-24 2013-03-11 Method for speech coding, method for speech decoding and their apparatuses Active US8688439B2 (en)
US20130204615A1 true US20130204615A1 (en) 2013-08-08
US8688439B2 true US8688439B2 (en) 2014-04-01
Canadian Office Action dated Feb. 21, 2013, issued in Canadian Patent Application No. 2,722,196 (2 pages).
European Examination Report dated Jan. 17, 2013, issued in European Application No. 09014422.1 (9 pages).
Gerson et al: "Vector Sum Excited Liner Prediction (VSELP) Speeach Coding at 8kbps", Proc. IEEE Int. Conf. Acoust., Speech and Signal Process, Apr. 1990, pp. 461-464.
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