Enhanced coding to improve coded communication signals

Details Enhanced coding to improve coded communication signals Enhanced coding to improve coded communication signals

1998-12-18

2001-01-30

Dorvil, Richemond (Department: 2741)

Data processing: speech signal processing, linguistics, language

Speech signal processing

For storage or transmission

C704S219000, C704S230000, CD14S358000, CD14S496000, C340S318000

Reexamination Certificate

active

06182030

ABSTRACT:

FIELD OF THE INVENTION
The invention relates generally to coding of signals in communication systems and, more particularly, to a feature for enhancement of coded communication signals.
BACKGROUND OF THE INVENTION
High quality coding of acoustical signals at low bit rates is of pivotal importance to communications systems such as mobile telephony, secure telephone, and voice storage. In recent years, there has been a strong trend in mobile telephony towards improved quality of the reconstructed acoustical signal and towards increased flexibility in the bit rate required for transmission. The trend towards improved quality reflects, on the one hand, the customer expectation that mobile telephony provides a quality equal to that of the regular telephone network. Particularly important in this respect is the performance for background signals and music. The trend towards flexibility in bit rate reflects, on the other hand, the desire of the service providers to operate near the network capacity without the risk of having to drop calls, and possibly to have different service levels with different cost. The ability to strip bits from an existing bit stream while maintaining the ability to reconstruct the speech signal (albeit at a lower accuracy) is an especially useful type of bit rate flexibility.
With existing speech coding technology, it is difficult to meet the simultaneous challenge of improved acoustic signal quality and increased flexibility in bit rate. This difficulty is the direct result of the structure of the linear-prediction based analysis-by-synthesis (LPAS) paradigm which is commonly used in mobile telephony. Currently, LPAS coders perform better in coding speech at rates between 5 and 20 kb/s than other technologies. Accordingly, the LPAS paradigm forms the basis of virtually every digital telephony standard, including GSM, D-AMPS, and PDC. However, while the performance for speech is good, current LPAS-based speech coders do not perform as well for music and background noise signals. Furthermore, the ability to strip bits from an existing bit stream until now implied the usage of relatively low efficiency algorithms.
The LPAS coding paradigm does not perform as well for non-speech sounds because it is optimized for the description of speech. Thus, the shape of the short-term power spectrum is described as the multiplication of a spectral envelope, which is described by an all-pole model (with almost always 10 poles), and the so-called spectral fine structure, which is a combination of two components which are harmonic and noise-like in character, respectively. In practice, it is found that this model is not sufficient for many music and background-noise signals. The model shortcomings manifest themselves in perceptually inadequate descriptions of the spectral valleys (zeros), peaks which are not part of the harmonic structure in an otherwise periodic signal, and a so-called “swirling” effect in steady background noise signals which is probably caused by the time variation in the parameter estimation error.
The two main existing approaches towards developing LPAS algorithms with increased flexibility in the bit rate have significant drawbacks. In the first approach, one simply combines a number of coders operating at different bit rates and selects one coder for a particular coding time segment (examples of this first approach are the TIA IS-95 and the more recent IS-127 standards). These types of coders will be referred to as “multi-rate” coders. The disadvantage of this method is that the signal reconstruction requires the arrival at the receiver of the entire bit stream of the selected coder. Thus, the bit stream cannot be altered after it leaves the transmitter.
In the second approach, embedded coding, the encoder produces a composite bit stream made up out of two or more separate bit streams: a primary bit stream which contains a basic description of the signal, and one or more auxiliary bit streams which contain information to enhance the basic signal description. In the LPAS setting, this second approach is implemented by a decomposition of the excitation signal of the LPAS coder into a primary excitation and one or more auxiliary excitations, which enhance the excitation. However, to maintain synchronicity between the encoder and decoder (fundamental for the LPAS paradigm) at all rates, the long-term predictor (present in virtually all LPAS paradigms) can only operate on the primary excitation. Since the long-term predictor provides the most significant part of the coding gain in the LPAS paradigm, this severely limits the benefit of the auxiliary excitations. Thus, these embedded LPAS coding algorithms provide increased bit rate flexibility at the expense of significantly curtailed coding efficiency.
For coders with fixed bit rates between 5 and 20 kb/s, the well-known LPAS paradigm dominates. Overviews of this coding paradigm are provided in, for example, P. Kroon and Ed. F. Deprettere, “A class of analysis-by-synthesis predictive coders for high quality speech coding at rates between 4.8 and 16 kbit/s”,
IEEE J. Selected Areas Comm.,
6:353-363, 1988; A. Gersho, “Advances in speech and audio compression”,
Proceedings IEEE,
82:900-918, 1994; and P. Kroon and W. B. Kleijn, “Linear-prediction based analysis-by-synthesis coding”, In W. B. Kleijn and K. K. Paliwal, editors,
Speech Coding and Synthesis,
pages 79-119. Elsevier Science Publishers, Amsterdam, 1995.
In the LPAS paradigm, the speech signal is reconstructed by exciting an adaptive synthesis filter with an excitation signal. The adaptive synthesis filter, which has an all-pole structure, is specified by the so-called linear prediction (LP) coefficients, which are adapted once per subframe (a subframe is typically 2 to 5 ms). The LP coefficients are estimated from the original signal once per frame (10 to 25 ms) and their value for each subframe is computed by interpolation. Information about the LP coefficients is usually transmitted once per frame. The excitation is the sum of two components: the adaptive-codebook (for the present purpose identical to the long-term predictor) contribution, and the fixed-codebook contribution.
The adaptive-codebook contribution is determined by selecting for the present subframe that segment of the past excitation which after filtering with the synthesis filter results in a reconstructed signal which is most similar to the original acoustic signal. The fixed-codebook contribution is the entry from a codebook of excitation vectors which, given the adaptive codebook contribution, renders the reconstructed signal obtained most similar to the original signal. In addition to the above process, the adaptive and fixed-codebook contributions are scaled by a quantized scaling factor.
The above description of the LPAS paradigm is applicable to almost all state-of-the-art coders. Examples of such coders are the 8 kb/s ITU G.729 (see R. Salami, C. Laflamme, J.-P. Adoul, and D. Massaloux, “A toll quality 8 kb/s speech codec for the personal communications system (PCS)”,
IEEE Trans. Vehic. Techn.,
43(3):808-816, 1994; and R. Salami et al., “Description of the proposed ITU-T 8 kb/s speech coding standard”,
Proc. IEEE Speech Coding Workshop,
pages 3-4, Annapolis, Md., 1995) and the GSM enhanced full-rate (GSMEFR) 12.2 kb/s coder (see European Telecommun. Standard Institute (ETSI), “Enhanced Full Rate (EFR) speech transcoding (GSM 06.60)”, ETSI Technical Standard 300 726, 1996). Both of these coders perform well for speech signals. However, for music signals both coders contain clearly audible artifacts, more so for the lower-rate coder. For each of these coders the entire bit stream must be obtained by the receiver to allow reconstruction.
The 16 kb/s ITU G.728 coder differs from the above paradigm outline in that the LP parameters are computed from the past reconstructed signal, and thus are not required to be transmitted. This is commonly referred to as backward LP adaptation. Only a fixed codebook is used. In contrast to other coders (which use a linear prediction order

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