## Linear Predictive Coding

Linear Predictive Coding is a method of using a constant number of known sample-values, that precede an unknown sample-value, and to find coefficients for each of the preceding sample-values, which they should be multiplied by, and the products summed, to derive the most-probable following sample-value.

More specifically, while the exercise of applying these coefficients should not require much explaining, methods for deriving them do.

Finding these coefficients is also called an Auto-Correlation, because the following sample is part of the same sequence, as the preceding samples belonged to.

Even though LPC travels along the stream of values, each preceding position relative to the current sample to predict is treated as having a persistent meaning, and for the sake of simplicity I’ll be referring to each preceding sample-position as One Predictor.

If the Linear Predictive Coding was only to be of the 2nd order, thus taking into account 2 Predictors, then it will often be simple to use a fixed system of coefficients.

In this case, the coefficients should be { -1, +2 }, which will successfully predict the continuation of a straight line, and nothing else.

One fact about a set of coefficients is, that their sum should be equal to 1, in order to predict a DC value correctly.

( If the intent was to use a set of 3 predictors, to conserve both the 1st and the 2nd derivatives of a curve, then the 3 coefficients should automatically be { +1, -3, +3 } . But, what’s needed for Signal Processing is often not, what’s needed for Analytical Geometry. )

But for orders of LPC greater than 3, the determination of the coefficients is anything but trivial. In order to understand how these coefficients can be computed, one must first understand a basic concept in Statistics called a Correlation. A correlation supposes that an ordered set of X-value and Y-value pairs exist, which could have any values for both X and Y, but that Y is supposed to follow from X, according to a linear equation, such that

Y = α + β X

Quite simply, the degree of correlation is the ideal value for β, which achieves the closest-fitting set of predicted Y-values, given hypothetical X-values.

The process of trying to compute this ideal value for β is also called Linear Regression Analysis, and I keep a fact-sheet about it:

This little sheet actually describes Non-Linear Regression Analysis at the top, using a matrix which states the polynomial terms of X, but it goes on to show the simpler examples of Linear Regression afterward.

There is a word of commentary to make, before understanding correlations at all. Essentially, they exist in two forms

1. There is a form, in which the products of the deviations of X and Y are divided by the variance of X, before being divided by the number of samples.
2. There is a form, in which the products of the deviations of X and Y are divided by the square root, of the product of the variance of X and the variance of Y, before being divided by the number of samples.

The variance of any data-set is also its standard deviation squared. And essentially, there are two ways to deal with the possibility of non-zero Y-intercepts – non-zero values of α. One way is to compute the mean of X, and to use the deviations of individual values of X from this mean, as well as to find the corresponding mean of Y, and to use deviations of individual values of Y from this mean.

Another way to do the Math, is what my fact-sheet describes.

Essentially, Form (1) above treats Y-values as following from known X-values, and is easily capable of indicating amounts of correlation greater than 1.

Form (2) finds how similar X and Y -values are, symmetrically, and should never produce correlations greater than 1.

For LPC, Form (2) is rather useless, and the mean of a set of predictors must be found anyway, so that individual deviations from this mean are also the easiest values to compute with.

The main idea when this is to become an autocorrelation, is that the correlation of the following sample is computed individually, as if it was one of the Y-values, as following each predictor, as if that was just one of the X-values. But it gets just a touch trickier…

(Last Edited 06/07/2017 … )

## One reason Why, It is Difficult For Me to Guess, at the Variable-Length Encoding of numbers, Chosen By Other People

The problem can come up often in Computing, that instead of having a fixed-length encoding for numbers, we may want to encode a majority of integers which lie in a small range, but out of which a progressively smaller number have larger values. This can lead to variable-length encoding schemes, and MP3 sound compression, logically, is one place where this happens.

If one is trying to guess at what encoding was used, the fact that can stymie a person is, that many methods exist to accomplish exactly that. Huffman Encoding has as a problem, that although the higher-value integers are assigned longer bit-sequences, the relative frequency with which these higher integers will occur, is not the inverse, of their bit-length. This can also be why, a non-default arrangement can be made, that if the size of the integer reaches 15, a full-length value needs to follow.

I have now finally learned, that with MP3 compression, at least the integers smaller than 15, Huffman-Encoded, are intended to be the default case, and values at or above 15 are intended to be the exception. Thus, if the scaling factor is increased, for sure the bit-length of the stream will decrease, until the bit-rate is achieved, that the user set. I got that.

But Why, then, You May Ask, does Dirk choose a different way to accomplish the same thing, and so often?

My answer would be, that formal solutions are often good at compressing the size of integers when those lie in a certain range, but if one value appears in the stream which is much larger than the average, those variable-length encoding schemes can become monsters. ( :1 )

As an example, I read that ‘FLAC’ will record the Linear Predictive Encoding coefficients for a frame accurately, and that this scheme will then Rice Encode the residual each time.

Pure Rice Encoding means, that a remainder of fixed bit-length is encoded with each sample, but that it is given a (variable-length) prefix encoded in straight unary, which states what the multiple of the tunable parameter is (the quotient), that fits into the fixed-length remainder. This choice of a pure unary prefix is questionable, for the reason that I just stated above.

Now, I know that there is also Exponential-Golomb encoding, which like Huffman Coding has a bit-length that grows with the size of the integer to be encoded. But Exponential-Golomb generally produces a bit-length twice as long, as what it would take just to write out the integer on paper.

And so at least a slightly more sophisticated form of encoding exists, which is called Golomb-Rice Encoding, which is essentially Rice Encoding, but in which the prefix, which states the quotient, is prepended in Exponential-Golomb format. Why would they not use it?

And, since it is possible just to put a prefix before an integer in unary, that states its length, an approach which I would be tempted to use, would be just to assume that this unary prefix should be multiplied by a factor such as 3, to arrive at the true length of the integer.

But then a problem with that would be, the fact that this type of prefix would need to be at least 2 bits long, for non-zero values, followed by this multiple of bits belonging to the value, as a minimum. So it will not compress very small values well.

And the reason for this would be the fact, that it would no longer be certain then, that the first bit which actually belongs to the integer, will always be a (1), the way it is with Exponential-Golomb.

And, while I tend to view such encoding schemes as arbitrary, the fashion these days is, always to select a formally-defined one.

In general, my approaches will work well, if a substantial number of values are high.

Dirk

1: ) And what ‘FLAC’ will do in such a case, is just switch the type of the frame this happens in, to a type ‘VERBATIM’ frame. In other words, FLAC would just decide ‘This is one frame we cannot compress’.

FLAC also has a mode, in which each sample is stated as a delta, from the previous one. This corresponds to an ‘LPE’ with one predictor, the coefficient of which is just equal to (+1), relative to which the current value is just the residual…

(Edit 05/25/2016 : ) This is another posting of mine, in which I explain an additional detail about MP3 compression.

Further, ‘FLAC’ is able to encode some of its frames as using LPE, with a variable number of coefficients. I.e., When set to compress more, it will spend its CPU time trying encodings with (!) 6 or more predictors, and will store those in cases where doing so led to more-compact encoding.

While a set of 4 or more coefficients needs to be computed specifically for one frame, via a Statistical Regression Analysis, I have read that for 1, 2, or 3, FLAC just uses a standard set of them. For 1, that will be [ +1 ] . For 2, that will be [ -1, +2 ] . It might seem like a purely academic exercise, to know a standard set of coefficients, which will generally work well if there are only 3 of them. But in fact, having this available offers a non-trivial advantage, over having to store those in the compressed stream.

Since we would presumably be multiplying signed, 16-bit samples with signed, 16-bit coefficients, it will be helpful if the latter only need to fall into the fractional range of ( -1.0 … +1.0 ) . The reason for this is the fact that If we needed to store coefficients which are allowed to exceed ( + 1.0 ) , Then we are blowing another bit of precision just so that one coefficient could do so.

The most recent coefficient will still exceed ( +1.0 ) when there are 3. But as soon as there are 4, none of them would exceed ( +1.0 ) anymore. Therefore, all the coefficients which must be stored, when their number reaches 4 or more, can be made more precise, just because there is a standard set for when we have 3.

## Sound Compression is also Possible, in the Time Domain.

One alternative type of sound compression which exists, is ‘FLAC’. This is a method which preserves all the information in the stream (later reproducing it exactly), which uses Linear Predictive Coding, and which Rice-Encodes the Residual, by which actual samples differ from the LPE-predicted values.

When I first tried to understand FLAC, my efforts were lacking, in that I did not know that LPE takes place in the time-domain, and not in the frequency-domain.

Dirk