Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission
Reexamination Certificate
2000-01-14
2002-06-18
Dorvil, Richemond (Department: 2654)
Data processing: speech signal processing, linguistics, language
Speech signal processing
For storage or transmission
C704S223000, C704S220000
Reexamination Certificate
active
06408267
ABSTRACT:
BACKGROUND OF THE INVENTION
The present invention concerns the field of digital coding of audio signals. It relates more particularly to a decoding method used to reconstitute an audio signal coded using a method employing a “backward LPC” synthesis filter.
Predictive block coding systems analyses successive frames of samples of the audio signal (generally speech or music) to be coded to extract a number of parameters for each frame. Those parameters are quantised to form a bit stream sent over a transmission channel.
Depending on the quality of the channel and the type of transport, the signal transmitted can be subject to interference causing errors in the bit stream received by the decoder. These errors in the bit stream can be isolated. However, they very frequently occur in bursts, especially in mobile radio channels with a high level of interference and in packet mode transmission networks. In this case, an entire packet of bits corresponding to one or more signal frames is erroneous or is not received.
The transmission system employed can frequently detect erroneous or missing frames at the level of the decoder. So-called “missing frame recovery” procedures are then used. These procedures enable the decoder to extrapolate the missing signal samples from samples recovered in frames preceding and possibly following the areas in which frames are missing.
The present invention aims to improve techniques for recovering missing frames in a manner that strongly limits subjective degradation of the signal perceived at the decoder in the presence of missing frames. It is of more particular benefit in the case of predictive coders using a technique generally known as “backward LPC analysis” continuously or intermittently. The abbreviation “LPC” signifies “linear predictive coding” and “backward” indicates that the analysis is performed on signals preceding the current frame. This technique is particularly sensitive to transmission errors in general and to missing frames in particular.
The most widely used linear prediction coding systems are CELP (Code-Excited Linear Predictive) coders. Backward LPC analysis in a CELP coder was used for the first time in the LD-CELP coder adopted by the ITV-T (see ITV-T Recommendation G.728). This coder can reduce the bit rate from 64 kbit/s to 16 kbit/s without degrading the perceived subjective quality.
Backward LPC analysis consists in performing the LPC analysis on the synthesised signal instead of on the current frame of the original audio signal. In reality, the analysis is performed on samples of the synthesised signal from frames preceding the current frame because that signal is available both at the coder (by virtue of local decoding that is generally useful in analysis-by-synthesis coders) and at the remote decoder. Because the analysis is performed at the coder and at the decoder, the LPC coefficients obtained do not have to be transmitted.
Compared to the more conventional “forward” LPC analysis, in which the linear prediction is applied to the signal at the input of the coder, backward LPC analysis provides a higher bit rate, which can be used to enrich the excitation dictionaries in the case of the CELP, for example. Also, and without increasing the bit rate, it significantly increases the order of analysis, the LPC synthesis filter typically having 50 coefficients for the LD-CELP coder as compared to 10 coefficients for most coders using forward LPC analysis.
Because of the higher order of the LPC filter, backward LPC analysis provides better modelling of musical signals, the spectrum of which is significantly richer than that of speech signals. Another reason why this technique is well suited to coding music signals is that music signals generally having a more stationary spectrum than speech signals, which improves the performance of backward LPC analysis. On the other hand, correct functioning of backward LPC analysis requires:
(i) A good quality synthesised signal, which must be very close to the original signal. This imposes a relatively high coding bit rate. Given the quality of current CELP coders, 13 kbit/s would seem to be the lower limit.
(ii) A short frame or a sufficiently stationary signal. There is a delay of one frame between the analysed signal and the signal to be coded. The frame length must therefore be short compared to the average time for which the signal is stationary.
(iii) Few transmission errors between the coder and the decoder. As soon as the synthesised signals are different, the coder and the decoder no longer calculate the same filter. Large divergences can then arise and be amplified, even in the absence of any new interference.
The sensitivity of backward LPC analysis coders/decoders to transmission errors is due mainly to the following recursive phenomenon: the difference between the synthesised signal generated at the coder (local decoder) and the synthesised signal reconstructed at the decoder by a missing frame recovery device causes a difference between the backward LPC filter calculated at the decoder for the next frame and that calculated at the coder, because these filters are calculated on the basis of the different signals. Those filters are used in turn to generate the synthesised signals of the next frame, which will therefore be different at the coder and at the decoder. The phenomenon can therefore propagate, increase in magnitude and cause the coder and decoder to diverge greatly and irreversibly. As backward LPC filters are generally of a high order (30 to 50 coefficients), they make a large contribution to the spectrum of the synthesised signal (high prediction gains).
Many coding algorithms use missing frame recovery techniques. The decoder is informed of a missing frame by one means or another (in mobile radio systems, for example, by receiving frame loss information from the channel decoder which detects transmission errors and can correct some of them). The objective of missing frame recovery devices is to extrapolate the samples of the missing frame from one or more of the most recent preceding frames which are deemed to be valid. Some systems extrapolate these samples using waveform substitution techniques which take samples directly from past decoded signals (see D. J. Goodman et al. : “Waveform Substitution Techniques for Recovering Missing Speech Segments in Packet Voice Communications”, IEEE Trans. On ASSP, Vol. ASSP-34, No.6, December 1986). In the case of predictive coders, of the CELP type, for example, the samples of missing frames are replaced using the synthesis model used to synthesise the valid frames. The missing frame recovery procedure must then supply the parameters needed for the synthesis which are not available for the missing frames (see, for example, ITV-T Recommendations G.723.1 and G.729). Some parameters manipulated or coded by predictive coders exhibit high correlation between frames. This applies in particular to LPC parameters and to long-term prediction parameters (LTP delay and associated gain) for voiced sounds. Because of this correlation, it is more advantageous to use the parameters of the last valid frame again to synthesise the missing frame rather than to use erroneous or random parameters.
For the CELP coding algorithm, the parameters of the missing frame are conventionally obtained in the following manner:
the LPC filter is obtained from the LPC parameters of the last valid frame, either by merely copying the parameters or introducing some damping;
voiced
on-voiced detection determines the harmonic content of the signal at the level of the missing frame (cf. ITV-T Recommendation G.723.1);
in the non-voiced situation, an excitation signal is generated in a partly random manner, for example by drawing a code word at random and using the past excitation gain slightly damped (cf. ITV-T Recommendation G.729), or random selection in the past excitation (cf. ITV-T Recommendation G.728);
in the case of a voiced signal, the LTP delay is generally that calculated in the preceding frame, possibly with slight “jitter” to prevent an excessively prol
Dorvil Richemond
France Telecom
Marshall Gerstein & Borun
LandOfFree
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