Source Coding

Source Coding

In a first-generation cellular handset, you talk into a microphone and a variable voltage
is produced, describing 3 kHz of voice modulated audio bandwidth. The voltage
is then FM-modulated onto an RF carrier�"an all analog processing chain.
In second-generation handsets, you talk into a microphone and the voice is turned
into a digital bit stream using waveform encoding. For example, in GSM, a 104 kbps
data stream is produced prior to the vocoder. It is the vocoder’s job to reduce this data
rate to, for example, 13 kbps or less without noticeable loss of quality. In the wireline
world and in digital cordless phones, this is achieved in the time domain by using time
domain compression techniques (exploiting sample-to-sample predictability). These
are known as adaptive differential pulse code modulation codecs. They work well in high
background noise conditions but suffer quality loss at low codec rates�"for example,
16 kbps or below.
The decision was made that digital cellular handsets should use speech synthesis
codecs that coded in the frequency domain (see Figure 1.11). The figure shows a female
voice saying “der.” Each block represents a 20-ms speech sample. The first block shows
the “d,” and the second block shows the “er” described in the time domain (y-axis) and
frequency domain (x-axis). Each sample is described in terms of frequency coefficients.
Compression is achieved by exploiting similarity between samples.

In the receiver, the frequency coefficients are used to rebuild, or synthesize, the harmonic
structure of the original voice sample. The more processing power used in the
codec, the better the quality for a given compression ratio.
Alternatively, rather than synthesize waveforms, waveforms can be prestored and
fetched and inserted as needed in the decoder. This reduces processor overhead but
increases memory bandwidth in the vocoder. These codecs are known as codebook
codecs or more precisely codebook excitation linear prediction (CELP) codecs. Codecs
used in present CDMA handsets and most future handsets are CELP codecs.
Voice codecs are also becoming variable rate, either switchable (for coverage or
capacity gain) or adaptive (the codec rate varies according to the dynamic range of the
input waveform). The objective of all codecs is to use processor bandwidth to reduce
transmission bandwidth. Speech synthesis codecs and codebook codecs can deliver
compression ratios of 8:1 or more without significant loss of quality.
3G handsets add in MPEG-4 encoders/decoders to support image and video processing.
In common with vocoders, these video codecs use time domain to frequency
domain transforms (specifically, a discrete cosine transform) to identify redundancy in
the input image waveform. As we will see, video codecs are capable of delivering compression
ratios of 40:1 or more with tolerable image quality.
Fourth-generation digital encoders will add in embedded rendering and mesh coding
techniques to support motion prediction, motion estimation, and motion compensation.