Signal Compression

Dr. Waquar Ahmad

National Institute of Technology Calicut

waquar@nitc.ac.in

Dr. Waquar Ahmad (NITC) Lecture-6 1 / 11

Subband Coding-Introduction
Temporal or spatial domain allowed us to use signal correlation properties to develop
techniques such as vector quantization.
Signals in the frequency domain allowed us to use their spectral structure and develop
transform coding approaches.
Now, we look at an approach that makes use of the structure in both domains.
In subband coding the signal is decomposed into frequency bands after which the
temporal or spatial signal in each band can be separately encoded.
The low frequency signals tend to be smoother allowing us to use sample to sample
correlation.
The high frequency signals tend to be sparse allowing us to use schemes more appropriate
to sparse signals.
Subband decomposition also permits the exploitation of both spectral and temporal
limitations of human perception which can let us selectively ignore or discard components
of the signal.
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Filters

A system that isolates certain frequency components is called a filter.

Filters that only let through components below a certain frequency f0 are called low-pass
filters.
filters that block all frequency components below a certain value f0 are called high-pass
filters.
The frequency f0 is called the cutoff frequency.
Filters that let through components that have frequency content above some frequency f1
but below frequency f2 are called band-pass filters.

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.
Digital filtering involves taking a weighted sum of current and past inputs to the filter
and, in some cases, the past outputs of the filter. The general form of the input–output
relationships of the filter is given by
N
X M
X
yn = ai xn−i + bi yn−i
i=0 i=0
where the sequence xn is the input to the filter, the sequence yn is the output from the
filter, and the values ai and bi are called the filter coefficients.
If the input sequence is a single 1 followed by all 0s, the output sequence is called the
impulse response of the filter.
If the bi are all 0, then the impulse response will die out after N samples −− > Finite
impulse response (FIR) filters.
The number N is sometimes called the number of taps in the filter.
If any of bi have nonzero values, the impulse response can continue forever−− > Infinite
impulse response (IIR) filters.
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Example
The most frequently used filter banks in subband coding consist of a cascade of stages,
where each stage consists of a low-pass filter and a high-pass filter, as shown in Fig

Dr. Waquar Ahmad (NITC) Lecture-6 5 / 11

QMF

The most popular among these filters are the quadrature mirror filters (QMF).
These filters have the property that if the impulse response of the low-pass filter is given
by hn , then the high-pass impulse response is given by (−1)n hN−1−n .
Filters with fewer taps are less efficient in their decomposition than the filters with more
taps.
However more taps leads to more computation.
Another popular set of filters are the Smith–Barnwell filters.
Cutoff for the Smith–Barnwell filter is much sharper than the cutoff for the QMF filter.
This means that the separation provided by the QMF filter is not as good as that
provided by the Smith–Barnwell filters.

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THE BASIC SUBBAND CODING ALGORITHM

The basic subband coding system is shown in Figure.

Figure: Block diagram of the subband coding system.

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Analysis

The source output is passed through a bank of filters, called the analysis filter bank.
The passbands of the filters can be nonoverlapping or overlapping.
The outputs of the filters are then subsampled.
we can reduce the number of samples at the output of the filter because the range of
frequencies at the output of the filter is less than the range of frequencies at the input to
the filter.
If the bandwidth at the output of the filter is 1/M of the bandwidth at the input to the
filter, we would decimate the output by a factor of M by keeping every M th sample.
Once the output of the filters has been decimated, the output is encoded using one of
several encoding schemes,for example vector quantization.

Dr. Waquar Ahmad (NITC) Lecture-6 8 / 11

QUANTIZATION AND CODING

Along with the selection of the compression scheme, the allocation of bits between the
subbands is an important design parameter.
Different subbands contain differing amounts of information. Therefore, we need to
allocate the available bits among the subbands according to its information content.
This bit allocation procedure can have a significant impact on the quality of the final
reconstruction, especially when the information content of different bands is very different.

Dr. Waquar Ahmad (NITC) Lecture-6 9 / 11

SYNTHESIS

The quantized and coded coefficients are used to reconstruct a representation of the
original signal at the decoder.
First, the encoded samples from each subband are decoded at the receiver.
These decoded values are then upsampled by inserting an appropriate number of 0s
between samples.
Once the number of samples per second has been brought back to the original rate, the
upsampled signals are passed through a bank of reconstruction filters.
The outputs of the reconstruction filters are added to give the final reconstructed outputs.

Dr. Waquar Ahmad (NITC) Lecture-6 10 / 11

Applications

Speech Coding
Audio Coding
Image Compression

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