Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission
Reexamination Certificate
2000-02-25
2002-07-23
Banks-Harold, Marsha D. (Department: 2654)
Data processing: speech signal processing, linguistics, language
Speech signal processing
For storage or transmission
C704S220000, C704S221000, C704S225000
Reexamination Certificate
active
06424940
ABSTRACT:
FIELD OF THE INVENTION
The present invention relates to telecommunication systems in general, and in particular to the transmission of compressed signals in telecommunications systems.
BACKGROUND OF THE INVENTION
In recent years, various techniques are being implemented in order to save on required bandwidth, techniques which achieve toll-quality or near toll-quality speech while using compressed telecommunication transmissions. These techniques typically involve the use of coding algorithms that allow reducing the bandwidth requirement of 64 kb/s for non-compressed transmissions. One such example is the LD-CELP algorithm that enables reducing the bandwidth requirement to 16 kb/s. Naturally, in order to use such coding algorithms, both ends of the transmission path must be provided with the ability to code and decode the transmissions. One solution for this requirement is using single proprietary equipment at both ends of and along the transmission path. Another possible solution is the implementation of international standards that allow compatibility of different types of equipment located along a transmission path.
The international standard for the coding algorithm LD-CELP was published on March 1995 as International Telecommunication Union (ITU-T) Recommendation G.728. However, it was found that this Recommendation contained several drawbacks. Among these drawbacks was the handling of transmissions at variable bit rate (referred to hereinafter as “VBR”). This problem was particularly noticed when G.728 Recommendation was used in voiceband data applications.
In its contribution to the ITU-T of Mar. 17, 1997, ECI Telecom Ltd. suggested a solution disclosed in Annex J of ITU-T Recommendation G. 728. The contribution, entitled “Variable Bit-Rate algorithm, mainly for the Voiceband data applications of LD-CELP ITU-T Rec. G. 728 in DCME” is hereby incorporated by reference. This publication will be referred to hereinafter as “40 kbps algorithm”.
In this contribution, a solution for VBR and particularly for voice-band data (to be referred to hereinafter as “VBD”) application, was described. The contribution provided information for the implementation of a codec that complies with the LD-CELP algorithm, as well as modification to Annex G of Rec. G 728, “16 kbit/s fixed point specification”, so as to enable a mode-switch on a fixed point arithmetic device.
The codec described in the 40 kbps algorithm basically uses a transmission rate of 40 kbit/s. The algorithmic delay is 5-samples long, totaling 0.625 msec, and the codec can perform a mode-switch every “adaptation-cycle” (2.5 msec).
The suggested 40 kbps algorithm, was intended mainly to solve problems in the transmission of compressed VBD for applications such as DCME, and was suggested to replace the 40 kbps ADPCM mode (ITU-T Rec.—G.726) in DCME systems where LD-CELP algorithm is incorporated. Among the features provided by this algorithm is the soft transition to and from the LD-CELP algorithm, and the maintaining of toll-quality or near toll-quality of speech.
The adaptation cycle used for the speech mode in the 40 kbps algorithm is essentially provided by G. 728 Recommendation. Therefore, when reverting to speech mode type of operation, the LD-CELP mode specified in Recommendation G.728 will be applied rather than the 40 kbps algorithm.
The main modification of a codec operating in accordance with the 40 kpbs algorithm is the implementation of the Trellis Coded Quantization (referred to hereinafter as “TCQ”) approach, described in IEEE Transactions on Communications Vol. 38, No. 1, (1990) which is hereby incorporated by reference. This TCQ approach replaces the analysis-by-synthesis approach to codebook search of ITU-T Rec. G. 728, in the VBD mode.
Still, in the 40 kbps algorithm suggested, no solution was provided to the problem of how to avoid reaching a saturation state when an impulse occurs in the prediction error, e.g. when having a sudden substantial change in the energy level of the prediction error. This problem results in generating a high level of noise at the output of the decoder, and is known to be a cause for discrepancies between the transmitting and the receiving ends of the transmission path.
U.S. Pat. No. 4,677,423 recognizes a somewhat similar problem associated with another type of algorithm, the ADPCM algorithm, and discloses a solution to that problem. The mechanism described in U.S. Pat. No. 4,677,423 is one for overcoming the problem associated with transitions in partial band energy signals, by locking and unlocking the adaptation speed. The adaptation speed is locked in cases of very slow speed of adaptation, while the unlocked mode is used when high speed of adaptation is required. Unfortunately, since this solution is not fast enough for systems having coding algorithms where the predictor is not an adaptive one, e.g. based on Linear Prediction (referred to hereinafter as LP) analysis, a different solution is required. A number of problems render the solution described in U.S. Pat. No. 4,677,423 inefficient when trying to avoid saturation in systems incorporating linear predictors, when an impulse occurs in the prediction error. Some of these problems are: the '423 solution is based on fact that each sample should be handled individually, whereas in linear predictors, a vector comprising a number of samples is used rather than single samples as suggested in the '423 solution, a difference which renders the '423 solution not fast enough to be applied in linear predictors systems. Another basic difference is, that the-errors handles by the '423 patent are logarithmic errors which are not likely to saturate the quantizer as fast as linear errors might. Therefore a different solution is required, one that can provide an answer to systems where linear predictors are incorporated.
SUMMARY OF THE INVENTION
It is therefore an object of the present invention to provide a method for determining the compensated scaling of a quantizer in a coder using a vectorial linear non-adaptive predicting algorithm, a method that overcomes the drawbacks of the prior art solutions described above.
It is another object of the present invention to provide a digital communication apparatus and system enabling to overcome problems caused by impulses occurring in the prediction error.
Further objects and features of the invention will become apparent from the following description and the accompanying drawings.
In accordance with the present invention there is provided a method for determining the compensated scaling of a quantizer in a process of encoding/decoding a VBD type transmission by using a vectorial linear non-adaptive predicting type algorithm.
The term “VBD” as will be referred hereinafter, is used to denote digital signals modulated for transmission in the voice band frequency (up to 4 KHz), e.g. modem signals, DTMF signals, or any other such narrow band type of signals.
The method provided by the present invention, preferably comprises the steps of:
i. providing a digital sample vector in a coded form;
ii. calculating LP coefficients for predicting said digital sample vector and deriving a linear prediction error vector therefrom;
iii. calculating the gain of said linear prediction error vector;
iv. calculating the scaling of the quantizer from said gain;
v. calculating an average value of said gain corresponding to said digital sample vector, based on preceding digital samples;
vi. calculating the difference between said gain and said average value;
vii. determining whether a gain compensation is required for an impulse in the prediction error of said digital sample vector, based on:
(a) comparing said difference with a first pre-defined threshold value, and
(b) comparing the differences between the gains associated with a pre-defined number of most recent digital sample vector provided and their corresponding average values and a second pre-defined threshold;
viii. in the case that the determination in step (vii) is that a gain compensation is required, determining the compensation requir
Agassy Meir
Ilan Amir
Banks-Harold Marsha D.
ECI Telecom Ltd.
McFadden Susan
Nath & Associates PLLC
Novick Harold L.
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