Audio Coding: An Introduction to Data Compression - Part 2

16.3.2 Layer II Coding
The Layer II coder provides a higher compression rate by making some relatively minor modifications to the Layer I coding scheme. These modifications include how the samples are grouped together, the representation of the scalefactors, and the quantization strategy.

Where the Layer I coder puts 12 samples from each subband into a frame, the Layer II coder groups three sets of 12 samples from each subband into a frame. The total number of samples per frame increases from 384 samples to 1152 samples. This reduces the amount of overhead per sample.

In Layer I coding a separate scalefactor is selected for each block of 12 samples. In Layer II coding the encoder tries to share a scale factor among two or all three groups of samples from each subband filter. The only time separate scalefactors are used for each group of 12 samples is when not doing so would result in a significant increase in distortion. The particular choice used in a frame is signaled through the scalefactor selection information field in the bitstream.

The major difference between the Layer I and Layer II coding schemes is in the quantization step. In the Layer I coding scheme the output of each subband is quantized using one of 14 possibilities; the same 14 possibilities for each of the subbands. In Layer II coding the quantizers used for each of the subbands can be selected from a different set of quantizers depending on the sampling rate and the bit rates.

For some sampling rate and bit rate combinations, many of the higher subbands are assigned 0 bits. That is, the information from those subbands is simply discarded. Where the quantizer selected has 3, 5, or 9 levels, the Layer II coding scheme uses one more enhancement.

Notice that in the case of 3 levels we have to use 2 bits per sample, which would have allowed us to represent 4 levels. The situation is even worse in the case of 5 levels, where we are forced to use 3 bits, wasting three codewords, and in the case of 9 levels where we have to use 4 bits, thus wasting 7 levels.

To avoid this situation, the Layer II coder groups 3 samples into a granule. If each sample can take on 3 levels, a granule can take on 27 levels. This can be accommodated using 5 bits. If each sample had been encoded separately we would have needed 6 bits. Similarly, if each sample can take on 9 values, a granule can take on 729 values. We can represent 729 values using 10 bits. If each sample in the granule had been encoded separately, we would have needed 12 bits. Using all these savings, the compression ratio in Layer II coding can be increase from 4:1 to 8:1 or 6:1.

The frame structure for the Layer II coder can be seen in Figure 16.6. The only real difference between this frame structure and the frame structure of the Layer I coder is the scalefactor selection information field.

16.3.3 Layer III Coding - mp3
Layer III coding, which has become widely popular under the name mp3, is considerably more complex than the Layer I and Layer II coding schemes. One of the problems with the Layer I and Layer II coding schemes was that with the 32-band decomposition, the bandwidth of the subbands at lower frequencies is significantly larger than the critical bands. This

makes it difficult to make an accurate judgement of the mask-to-signal ratio. If we get a high amplitude tone within a subband and if the subband was narrow enough, we could assume that it masked other tones in the band. However, if the bandwidth of the subband is significantly higher than the critical bandwidth at that frequency, it becomes more difficult to determine whether other tones in the subband will be be masked.

A simple way to increase the spectral resolution would be to decompose the signal directly into a higher number of bands. However, one of the requirements on the Layer III algorithm is that it be backward compatible with Layer I and Layer II coders. To satisfy this backward compatibility requirement, the spectral decomposition in the Layer III algorithm is performed in two stages.

First the 32-band subband decomposition used in Layer I and Layer II is employed. The output of each subband is transformed using a modified discrete cosine transform (MDCT) with a 50% overlap. The Layer III algorithm specifies two sizes for the MDCT, 6 or 18. This means that the output of each subband can be decomposed into 18 frequency coefficients or 6 frequency coefficients.

The reason for having two sizes for the MDCT is that when we transform a sequence into the frequency domain, we lose time resolution even as we gain frequency resolution. The larger the block size the more we lose in terms of time resolution. The problem with this is that any quantization noise introduced into the frequency coefficients will get spread over the entire block size of the transform. Backward temporal masking occurs for only a short duration prior to the masking sound (approximately 20 msec). Therefore, quantization noise will appear as a pre-echo.