Coded data generation or conversion – Analog to or from digital conversion – Analog to digital conversion
Patent
1993-04-09
1995-09-19
Young, Brian K.
Coded data generation or conversion
Analog to or from digital conversion
Analog to digital conversion
341106, H03M 112
Patent
active
054519519
DESCRIPTION:
BRIEF SUMMARY
TECHNICAL FIELD
The present invention relates to a method of, and system for, the coding of analogue signals having particular, but not exclusive, application to a low bit-rate speech coder for use in telecommunications. The present invention also relates to a corresponding method of, and system for, re-synthesising a perceptually close replica of the analogue signals originally coded.
BACKGROUND ART
For convenience of description the invention will be described with reference to a speech codec (coder-decoder) but the invention is also applicable to the coding and re-synthesising of other types of analogue signals, for example video. Digital techniques for the coding of speech are growing in popularity for a number of reasons, notably flexibility, cost and robustness to noise. One such technique is called Code Excited Linear Prediction (CELP) wherein the incoming speech signal is sampled, segmented into frames and encoded using a process which involves comparing it with sequences taken from a known codebook. The index number of the codebook sequence that provides the best match to each frame of the incoming speech is then stored or transmitted together with some gain and filter parameters. This type of coder belongs to the class of analysis by synthesis coders, so named since they synthesise a large number of possible matches for the signal to be coded and then use comparison techniques to analyse the incoming signal. The corresponding decoder or re-synthesiser will generally include a synthesis section similar to that of the coder.
"Fast CELP coding based on algebraic codes" by J-P. Adoul, P. Mabilleau, M. Delprat and S. Morissette, read at the International Conference on Acoustics, Speech and Signal Processing (ICASSP) 1987, pages 1957-1960 discloses a simple CELP speech coding system which is described briefly here.
The output of a source of original speech is fed to a sampling and segmentation means which quantises the speech at an appropriate sampling rate such as 8 kHz and segments it into frames with a length of, for example, 5 ms. The output of the segmentation means comprises sampled, segmented speech which is fed to a non-inverting input of a summer and to a Linear Predictive Coder (LPC). The LPC derives a set of filter coefficients relating to the short term redundancy in the incoming speech signal.
A two-dimensional codebook contains K stochastic sequences of sampled white Gaussian noise, each of length N samples. The frames of sampled speech from the segmentation means also have a length of N samples. The codebook sequences are referred to as c.sup.k (n), where k is the codebook index and n is the particular sample number within a given sequence number k. The selected output sequence c.sup.k (n) is fed to a gain stage having a gain G which gain is derived mathematically for each block of the sampled speech and each codebook sequence. The output of the gain stage is filtered successively in a long term filter and a short term filter. The long term filter usually has only one tap and a relatively long delay that is usually greater than the length of the frames of sampled speech. The purpose of the long term filter is to impose some long term order upon the codebook sequence and since the frequency of this long term order is more often than not the pitch of the speech being synthesised, this filter is also referred to as the pitch predictor. The transfer function of the long term filter is 1/B(z) and the filter coefficient may be derived by an adaptive loop or by analysis of the incoming speech signal. The short term filter has much shorter delays but a much larger number of taps (typically 10 to 20) than the long term filter. The purpose of the short term filter is to impose some short term order upon the codebook sequence which results, in real speech, from the speaker's vocal tract and so this filter is often referred to as the vocal tract filter. The transfer function of this filter is 1/A(z) and the filter coefficients are supplied to the filter by the LPC. The output of the short term f
REFERENCES:
patent: 4963034 (1990-10-01), Cuperman et al.
patent: 4979213 (1990-12-01), Nitta
patent: 5007092 (1991-04-01), Galand et al.
Adoul, P. Mabi Lleau et al., "Fast CELP Coding Based on Alegebraic Codes", International Conference on Acoustics, 1987, pp. 1957-1960.
Elliott Patrick W.
Moulsley Timothy J.
Treacy David R.
U.S. Philips Corporation
Young Brian K.
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