Information about Code Excited Linear Prediction

Code Excited Linear Prediction (CELP) is a speech coding algorithm originally proposed by M.R. Schroeder and B.S. Atal in 1985. At the time, it provided significantly better quality than existing low bit-rate algorithms, such as RELP and LPC vocoders (e.g. FS-1015). Along with its variants, such as ACELP, RCELP, LD-CELP and VSELP, it is currently the most widely used speech coding algorithm. CELP is now used as a generic term for a class of algorithms and not for a particular codec.

Introduction

The CELP algorithm is based on four main ideas:

• Using the source-filter model of speech production through linear prediction (LP);
• Using an adaptive and a fixed codebook as the input (excitation) of the LP model;
• Performing a search in closed-loop in a “perceptually weighted domain”.
• Applying vector quantization (VQ)

The original algorithm as proposed by Schroeder and Atal required 100 seconds to encode 1 second of speech when run on a Cray I supercomputer. Since then, more efficient ways of implementing the codebooks and improvements in computing capabilities have made it possible to run the algorithm in embedded devices, such as mobile phones.

CELP Decoder

Before exploring the complex encoding process of CELP we introduce the Speex decoder here. Figure 1 describes a generic CELP decoder. The excitation is produced by summing the contributions from an adaptive (aka pitch) codebook and a fixed (aka innovation) codebook:

where is the adaptive (pitch) codebook contribution and is the fixed (innovation) codebook contribution. The fixed codebook is a vector quantization dictionary that is (implicitly or explicitly) hard-coded into the codec. This codebook can be algebraic (ACELP) or be stored explicitly (e.g. Speex). The entries in the adaptive codebook consist of delayed versions of the excitation. This makes it possible to efficiently code periodic signals, such as voiced sounds.

The filter that shapes the excitation has an all-pole model of the form , where is called the prediction filter and is obtained using linear prediction (Levinson-Durbin algorithm). An all-pole filter is used because it is a good representation of the human vocal tract and because it is easy to compute.

CELP Encoder

The main principle behind CELP is called Analysis-by-Synthesis (AbS) and means that the encoding (analysis) is performed by perceptually optimising the decoded (synthesis) signal in a closed loop. In theory, the best CELP stream would be produced by trying all possible bit combinations and selecting the one that produces the best-sounding decoded signal. This is obviously not possible in practice for two reasons: the required complexity is beyond any currently available hardware and the "best sounding" selection criterion implies a human listener.

In order to achieve real-time encoding using limited computing resources, the CELP search is broken down into smaller, more manageable, sequential searches using a simple perceptual weighting function. Typically, the encoding is performed in the following order:

• LPC coefficients are computed and quantized, usually as LSPs
• The adaptive (pitch) codebook is searched and its contribution removed
• The fixed (innovation) codebook is searched

Noise Weighting

Most (if not all) modern audio codecs attempt to shape the coding noise so that it appears mostly in the frequency regions where the ear cannot detect it. For example, the ear is more tolerant to noise in parts of the spectrum that are louder and vice versa. That's why instead of minimizing the simple quadratic error, CELP minimizes the error for the perceptually weighted domain. The weighting filter W(z) is typically derived from the LPC filter by the use of bandwidth expansion:

where .

External links

• This is based on a paper presented at Linux.Conf.Au
• Some parts based on the Speex codec manual
• reference implementations of CELP 1016A and LPC 10e.

References

M. R. Schroeder and B. S. Atal, "Code-excited linear prediction (CELP): high-quality speech at very low bit rates," in Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP), vol. 10, pp. 937-940, 1985.