Electrical audio signal processing systems and devices – One-way audio signal program distribution – Public address system
Patent
1989-08-14
1991-10-15
Shaw, Dale M.
Electrical audio signal processing systems and devices
One-way audio signal program distribution
Public address system
381 36, G10L 302, G10L 914
Patent
active
050581651
DESCRIPTION:
BRIEF SUMMARY
BACKGROUND OF THE INVENTION
1. Field of the Invention
The invention is concerned with speech coding, and more particularly to systems in which a speech signal can be generated by feeding the output of an excitation source through a synthesis filter. The coding problem then becomes one of generating, from input speech, the necessary excitation and filter parameters. LPC (linear predictive coding) parameters for the filter can be derived using well-established techniques, and the present invention is concerned with the excitation source.
2. Related Art
Systems in which a voiced/unvoiced decision on the input speech is made to switch between a noise source and a repetitive pulse source tend to give the speech output an unnatural quality, and it has been proposed to employ a single "multipulse" excitation source in which a sequence of pulses is generated, no prior assumptions being made as to the nature of the sequence. It is found that, with this method, only a few pulses (say 8 in a 10 ms frame) are sufficient for obtaining reasonable results. See B S Atal and J R Remde: "A New Model of LPC Excitation for producing Natural-sounding Speech at Low Bit Rates", Proc. IEEE ICASSP, Paris, pp. 614, 1982.
SUMMARY OF THE INVENTION
According to the present invention there is provided a speech coder comprising means for deriving, from an input speech signal, parameters of a synthesis filter; means for generating a coded representation of an excitation consisting of a plurality of pulses within a time frame corresponding to a larger plurality of speech samples, being arranged in operation to select the amplitudes and timing of pulses so as to reduce the difference between the input speech signal and the response of the filter to the excitation by: an excitation tending to reduce the said difference, and successively deriving one or more further pulses which in combination with the first and any intervening pulses represent an excitation tending to reduce the said difference; their position in the derivation sequence; and a backward adaptive quantizer for quantizing the products.
BRIEF DESCRIPTION OF THE DRAWING
Some embodiments of the invention will now be described with reference to the accompanying drawings, in which:
FIG. 1 is a block diagram of one embodiment of speech coder; FIG. 2 is a block diagram of a decoder for use with the coder of FIG. 1; and
FIG. 3 is a block diagram of a second embodiment of coder.
DESCRIPTION OF PREFERRED EMBODIMENTS
In the coder of FIG. 1, input speech signals, in sampled (preferably digital) form at an input 1 are processed by a predictor 2 to produce an output (e.g. in the form of a set of filter coefficients) defining a synthesis filter having a spectral response akin to that of the speech signals. The predictor analysis can be any of those conventionally used in so-called LPC (linear predictive coding) speech coders. As in common in such systems, the analysis is performed on frames of speech into which the input samples are divided. Typically the frame length may be 20 mns; hence a set of coefficients is produced every 20 mns and supplied via lines 3 to an output multiplexer 4.
As well as the filter representation, the coder also produces a representation of an excitation which is to be generated at the decoder to drive the synthesis filter in order to produce an approximation to the original speech. The coder of FIG. 1 has a multipulse derivation unit 5 which derives from the input speech samples and the LPC coefficients the amplitudes (on output 6) and positions (on output 7) of the pulses in a "multipulse" excitation frame as mentioned above. Whilst the typical sub-block (i.e. portion of LPC frame) size of 10 ms with eight pulses may be employed, the embodiment of FIG. 1 employs a sub-block duration of 4 ms, with three pulses. This is preferred as introducing less delay into the coding process. The object of the multipulse derivation is to find the pulse positions and amplitudes which minimize the error between the decoded synthetic speech and the original speech.
If it
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Atal et al, IEEE International Conference on Acoustics, Speech and Signal Processing, vol. 1, May 1982, "A New Model of LPC Excitation for Producing Natural-Sounding Speech at Low Bit Rates", pp. 614-617.
Singhal et al., IEEE, "Improving Performance of Multi-Pulse LPC Coders at Low Bit Rates", 1984, pp. 1.3.1-1.3.4.
ICASSP '84 IEEE International Conference on Acoustics, Speech & Signal Processing, Mar. 19-21, 1984, San Diego, U.S., vol. 1, IEEE (New York, U.S.) M. Berouti et al: "Efficient Computation and Encoding of the Multipulse Excitation for LPC", pp. 10.1.1-10.1.4.
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British Telecommunications public limited company
Doerrler Michelle
Shaw Dale M.
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