Abstract
In the past, high-quality speech used to be obtainable only by high-bit-rate coders such as 64 kb/s log PCM or 32 kb/s ADPCM. Currently, several coding techniques can produce high-quality speech at 16 kb/s. These techniques include Code-Excited Linear Prediction (CELP) [1], Multi-Pulse Linear Predictive Coding (MPLPC) [2], Adaptive Predictive Coding (APC) [3], Adaptive Transform Coding (ATC) [4], and Sub-Band Coding (SBC) combined with ADPCM [5], etc. However, all of them require a large coding delay — typically 40 to 60 ms — to achieve high-quality speech. While a large delay is necessary for these coders to buffer enough speech to exploit the redundancy, the delay is undesirable in many applications, especially when echo cancellation is involved. Achieving high-quality speech at 16 kb/s with a coding delay less than 1 or 2 ms has been a major challenge to speech researchers.
This is a preview of subscription content, log in via an institution to check access.
Access this chapter
Tax calculation will be finalised at checkout
Purchases are for personal use only
Preview
Unable to display preview. Download preview PDF.
References
M. R. Schroeder and B. S. Atal, “Code-Excited Linear Prediction (CELP): high quality speech at very low bit rates,” Proc. IEEE Int. Conf. Acoust., Speech, Signal Processing, pp. 937–940 (1985).
B. S. Atal and J. R. Remde, “A new model of LPC excitation for producing natural-sounding speech at low bit rates,” Proc. IEEE Int. Conf. Acoust., Speech, Signal Processing, Paris, France, pp. 614–617 (April 1982).
B. S. Atal, “Predictive coding of speech at low bit rates,” IEEE Trans. Communications COM-30(4), pp. 600–614 (April 1982).
R. Zelinski and P. Noll, “Adaptive transform coding of speech signals,” IEEE Trans. Acoust., Speech, Signal Processing ASSP-25, pp. 299–309 (1977).
F. K. Soong, R. V. Cox, and N. S. Jayant, “A high quality subband speech coder with backward adaptive predictor and optimal time-frequency bit assignment,” Proc. IEEE Int. Conf. Acoust., Speech, Signal Processing, pp. 2387–2390 (1986).
CCITT Study Group XVIII, Terms of reference of the ad hoc group on 16 kbitls speech coding (Annex 1 to question U/XV), June 1988.
N. S. Jayant and V. Ramamoorthy, “Adaptive postfiltering of 16 kb/s-ADPCM speech,” Proc. IEEE Int. Conf. Acoust., Speech, Signal Processing, pp. 829–832 (1986).
M. Berouti, J. Jachner, D. Sloan, and P. Mermelstein, “Reducing signal delay in multi-pulse coding at 16 kb/s,” Proc. IEEE Int. Conf. Acoust., Speech, Signal Processing, pp. 3043–3046 (1986).
T. Taniguchi and et al., “A 16 kbps ADPCM with multi-quantizer (ADPCM-MQ) codec and its implementation by digital signal processor,” Proc. IEEE Int. Conf. Acoust., Speech, Signal Processing, pp. 1340–1343 (1987).
R. V. Cox, S. L. Gay, Y. Shoham, S. R. Quackenbush, N. Seshadri, and N. S. Jayant, “New directions in subband coding,” IEEE J. Selected Areas Comm. 6(2), pp. 391–409 (February 1988).
J. D. Gibson and G. B. Haschke, “Backward adaptive tree coding of speech at 16 kbps,” Proc. IEEE Int. Conf. Acoust., Speech, Signal Processing, New York, pp. 251–254 (April 1988).
V. Iyengar and P. Kabal, “A low delay 16 kbits/sec speech coder,” Proc. IEEE Int. Conf. Acoust., Speech, Signal Processing, New York, pp. 243–246 (April 1988).
L. Watts and V. Cuperman, “A Vector ADPCM analysis-by-synthesis configuration for 16 kbit/s speech coding,” Proc. IEEE Global Commun. Conf., pp. 275–279 (December 1988).
L. Cellario, G. Ferraris, and D. Sereno, “A 2 ms delay CELP coder,” Proc. IEEE Int. Conf. Acoust., Speech, Signal Processing,, pp. 73–76 (May 1989).
J.-H. Chen and A. Gersho, “Real-time vector APC speech coding at 4800 bps with adaptive postfiltering,” Proc. IEEE Int. Conf. Acoust., Speech, Signal Processing, pp. 2185–2188 (1987).
AT&T contributions to CCITT Study Group XV and T1Y1.2, October 1988–November 1989.
J.-H. Chen and A. Gersho, “Gain-adaptive vector quantization with application to speech coding,” IEEE Trans. Comm, pp. 918–930 (September 1987).
T. P. Bamwell, III., “Recursive windowing for generating autocorrelation coefficients for LPC analysis,” IEEE Trans. Acoust., Speech, Signal Processing ASSP-29(5), pp. 1062–1066 (October 1981).
N. S. Jayant, “Adaptive quantization with a one word memory,” Bell Syst. Tech. J. 52, pp. 1119–1144 (September 1973).
DJ. Goodman and R. M. Wilkinson, “A robust adaptive quantizer,” IEEE Trans. Comm., pp. 1362–1365 (November 1975).
I. M. Trancoso and B. S. Atal, “Efficient procedures for finding the optimum innovation in stochastic coders,” Proc. IEEE Int. Conf. Acoust., Speh, Signal Processing, pp. 2375–2379 (1986).
J. R. B. De Marca and N. S. Jayant, “An algorithm for assigning binary indices to the codevectors of a multi-dimensional quantizer,” Proc. IEEE Int. Conf. on Communications, pp. 1128–1132 (June 1987).
K. A. Zeger and A. Gersho, “Zero redundancy channel coding in vector quantization,” Electronics Letters 23(12), pp. 654–656 (June 1987).
Author information
Authors and Affiliations
Editor information
Editors and Affiliations
Rights and permissions
Copyright information
© 1991 Springer Science+Business Media New York
About this chapter
Cite this chapter
Chen, JH. (1991). A Robust Low-Delay CELP Speech Coder at 16 Kb/s. In: Atal, B.S., Cuperman, V., Gersho, A. (eds) Advances in Speech Coding. The Springer International Series in Engineering and Computer Science, vol 114. Springer, Boston, MA. https://doi.org/10.1007/978-1-4615-3266-8_4
Download citation
DOI: https://doi.org/10.1007/978-1-4615-3266-8_4
Publisher Name: Springer, Boston, MA
Print ISBN: 978-1-4613-6437-5
Online ISBN: 978-1-4615-3266-8
eBook Packages: Springer Book Archive