Audio coder/decoder with recursive determination of prediction c

Pulse or digital communications – Bandwidth reduction or expansion – Pulse code modulation

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375242, 348409, 395 228, 395 209, G01L 914, H03H 1700

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active

056573502

DESCRIPTION:

BRIEF SUMMARY
RELATED APPLICATION

This application is related to concurrently filed co-pending application Ser. No. 08/362,515, filed on Jan. 5, 1995, of the same inventor, now still pending.
The related application relates to a linear predictive coder or decoder wherein the number of recursive recomputations which must be repeated when an overflow occurs is minimized by only performing such repeated recomputation starting with the prediction coefficient at which overflow occurs. That is feasible because the values of the previously recomputed coefficients will correspond to the values that would be obtained if they were again recomputed using a revised block floating point format. The present Application relates to a predictive coder or decoder wherein recursive recomputations of prediction coefficients is performed more accurately but without increasing the complexity of the required calculations.


BACKGROUND OF THE INVENTION

1. Field of the Invention
The invention relates to a transmission system comprising a coder for coding a signal, and a transmitter for transmitting the coded signal to a receiver, which includes a decoder for decoding the coded signal, and an adaptive prediction filter.
The invention likewise relates to a terminal unit, a coder, a decoder and an adaptive prediction filter.
2. Description of the Related Art
Such a transmission system can be obtained, for example, without difficulty from the document "Draft text for Annex A 16 k/bits of G.728 Fixel Point Specification" Temporary Document 41(P), CCIT Study Group 15, Geneva, Sept. 7-17, 1993, which corresponds to no longer available Doc. AH. 93-D.3, CCITT Study Group XV, London, Mar. 29-30, 1993, and from CCITT Recommendation G.728, "Coding of speech signals at 16 kbit/s using low-delay code excited linear prediction," Geneva, 1992. The first two documents will henceforth be referenced [I] and [II]. Such transmission systems can be used to provide multiple use of a given transmission capacity. The reduction of the bit rate of a signal 64 coding makes it possible, for example, to hold four 16 k/bits telephone conversations simultaneously over one 64 kbit/s transmission channel.
The multiple use of a given transmission capacity has very much importance in radio channels, as will be readily understood if one considers of the rising number of subscribers of mobile radio systems. Also the storage capacity of a memory with an arbitrary storage medium can be used to considerably more advantage when bit rate reducing coders and appropriate decoders are used, because less storage space is necessary for storing an information signal.
It is known to utilize linear prediction for bit rate reduction. In [I] and [II] prediction coefficients are computed by an adaptive prediction filter. The computation is performed segment by segment using sample values of an auxiliary signal which may be an (electric) audio signal or speech signal, for example, coming from a person. Alternatively, it is possible for the auxiliary signal to be a synthetic audio or speech signal as produced in a coder which operates according to the principle of "analysis by synthesis". A linear correlation between a predicted sampling value (prediction value) of the auxiliary signal and previous sampling values of this signal is realised with the prediction coefficients. The prediction coefficients are determined so that the sum of the squares of all the errors computed for a segment of sampling values assumes a minimum value. An error is here meant to be understood as the difference between a sampling value and its predicted value. More accurate descriptions will be given hereinbelow.
In [I] and [II] an excited signal is converted into a synthetic audio signal by a synthesis filter. This synthetic audio signal is subtracted from an audio signal to be coded and the difference is used for optimizing the selection of the excited signal.
The computation of the prediction coefficients calls for correlation coefficients which are derived from the sampling values of the synthetic audio signal. The co

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Juin-Hwey Chen, "High-Quality 16 KB/S Speech Coding With a One-Way Delay Less Than 2 MS", pp. 453-456, 1990 IEEE S9.1.
L.R. Rabiner/R.W. Schafer, "Levinson-Durban Recursion" in the Textbook Digital Processin Go Fspeech Signals, Prentice Hall Signal Processing Series, pp. 396-421.
CCITT Recommendation G. 728, "Coding of Speech Signals at 16 K Bits/S".

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