Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission
Patent
1997-08-04
2000-01-04
Hudspeth, David R.
Data processing: speech signal processing, linguistics, language
Speech signal processing
For storage or transmission
704220, 704201, G10L 914, G10L 300
Patent
active
06012024&
DESCRIPTION:
BRIEF SUMMARY
TECHNICAL FIELD OF THE INVENTION
The invention is related to speech coding techniques and general speech processing. More particularly, it is related to speech coding methods based on analysis by synthesis schemes in combination with backward adaptation techniques.
DESCRIPTION OF RELATED ART
A system based on analysis by synthesis and backward adaptation is used for instance in the Low-Delay Code Excited Linear Prediction (LD-CELP) speech codec, that was recently standardised by the International Telecommunication Union (ITU) in the publication "CODING OF SPEECH AT 16 kbit/s USING LOW-DELAY CODE EXCITED LINEAR PREDICTION", copyright by ITU 1992, recommendation G.728. This speech signal compression algorithm is meanwhile well known under the speech coding experts all over the world.
Digital networks are used to transmit digitally encoded signals. In the past mainly speech signals were to be transmitted. Now the data traffic caused by a wide spread use of electronic mailing networks is worldwide growing more and more. From an economical stand point, the number of connected users must be maximized without network congestion. As a consequence, speech compression algorithms have been developed specially optimized by utilizing noise masking effects. Unfortunately, these coding algorithms are not well suited for the transmission of voiceband data signals. So the idea is to add signal classification algorithms and to use a suitable voiceband data signal compression algorithm, hereinafter referred to as VDSC algorithm when data signals are detected. Currently a 16 kb/s-Digital-Circuit-Multiplication-Equipment (DCME) transmission system is being standardised using this idea. The LD-CELP codec will be used for transmission of speech whereas for voiceband data transmission a new coding algorithm is being under development within ITU.
In practical applications the signal classification algorithm may fail resulting in more or less frequent switching between different coding schemes. If the next coding scheme would always start from the reset state this may not be critical during transmission of voice band data. However, when speech is currently being transmitted this would result in rather annoying effects.
In order to overcome this problem in 16 kb/s DCME systems it was proposed to keep the LD-CELP architecture also for voiceband data signal compression. Only the bit rate should be increased for example by providing larger shape codebooks in order to ensure sufficient quantization. With such a method a continuous shape of the time signal would be guaranteed when switching from one coding mode to the other.
The drawback of this solution is twofold: on the one hand, the computational load would be increased significantly during transmission at higher bit rates. This makes implementations not very attractive as the conventional LD-CELP requires nearly the complete computation power of digital signal processors (DSPs) which are currently offered on the market. On the other hand, it is very likely that the coding of voiceband data signals can be done much more efficiently with specially optimised architectures resulting in bit rates below 40 kb/s or higher performance. Hitherto, 40 kb/s seems to be the required bitrate for compressions algorithms used with voiceband data signals. It is trivial to mention that this switching problem also arises if already existing signal compression algorithms are used in combination with LD-CELP type codecs. Known systems are using for example the algorithms according to ITU rec. G.711 (64 kb/s) or G.726 (32 kb/s or 10 kb/s) when voice band data signals have to be transmitted.
In this connection a coding algorithm, named ADPCM, the structure of which has similarities to the LD-CELP in that it includes forward error correction. Reference is made to the document "Digital Communications" by Simon Haykin, John Wiley & Sons, 1988.
In the patent U.S. Pat. No. 5,233,660 is disclosed a low-delay digital speech encoder and decoder based on code excited linear prediction (LD-CELP). The coding inc
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Azad Abul K.
Hudspeth David R.
Telefonaktiebolaget LM Ericsson
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