Waveform Codecs

Waveform codecs attempt, without using any knowledge of how the signal to be coded was generated, to produce a reconstructed signal whose waveform is as close as possible to the original. This means that in theory they should be signal independent and work well with non-speech signals. Generally they are low complexity codecs which produce high quality speech at rates above about 16 kbits/s. When the data rate is lowered below this level the reconstructed speech quality that can be obtained degrades rapidly.

The simplest form of waveform coding is Pulse Code Modulation (PCM), which merely involves sampling and quantizing the input waveform. Narrow-band speech is typically band-limited to 4 kHz and sampled at 8 kHz. If linear quantization is used then to give good quality speech around twelve bits per sample are needed, giving a bit rate of 96 kbits/s. This bit rate can be reduced by using non-uniform quantization of the samples. In speech coding an approximation to a logarithmic quantizer is often used. Such quantizers give a signal to noise ratio which is almost constant over a wide range of input levels, and at a rate of eight bits/sample (or 64 kbits/s) give a reconstructed signal which is almost indistinguishable from the original. Such logarithmic quantizers were standardised in the 1960's, and are still widely used today. In America u-law companding is the standard, while in Europe the slightly different A-law compression is used. They have the advantages of low complexity and delay with high quality reproduced speech, but require a relatively high bit rate and have a high susceptibility to channel errors.

A commonly used technique in speech coding is to attempt to predict the value of the next sample from the previous samples. It is possible to do this because of the correlations present in speech samples due to the effects of the vocal tract and the vibrations of the vocal cords, as discussed earlier. If the predictions are effective then the error signal between the predicted samples and the actual speech samples will have a lower variance than the original speech samples. Therefore we should be able to quantize this error signal with fewer bits than the original speech signal. This is the basis of Differential Pulse Code Modulation (DPCM) schemes - they quantize the difference between the original and predicted signals.

The results from such codecs can be improved if the predictor and quantizer are made adaptive so that they change to match the characteristics of the speech being coded. This leads to Adaptive Differential PCM (ADPCM) codecs. In the mid 1980's the CCITT standardised a ADPCM codec operating at 32 kbits/s, which gave speech quality that was very similar to the 64 kbits/s PCM codecs. Later ADPCM codecs operating at 16,24 and 40 kbits/s were also standardised.

The waveform codecs described above all code speech with an entirely time domain approach. Frequency domain approaches are also possible, and have certain advantages. For example in Sub-Band Coding (SBC) the input speech is split into a number of frequency bands, or sub-bands, and each is coded independently using for example an ADPCM like coder. At the receiver the sub-band signals are decoded and recombined to give the reconstructed speech signal. The advantages of doing this come from the fact that the noise in each sub-band is dependent only on the coding used in that sub-band. Therefore we can allocate more bits to perceptually important sub-bands so that the noise in these frequency regions is low, while in other sub-bands we may be content to allow a high coding noise because noise at these frequencies is less perceptually important. Adaptive bit allocation schemes may be used to further exploit these ideas. Sub-band codecs tend to produce communications to toll quality speech in the range 16-32 kbits/s. Due to the filtering necessary to split the speech into sub-bands they are more complex than simple DPCM coders, and introduce more coding delay. However the complexity and delay are still relatively low when compared to most hybrid codecs.

Another frequency domain waveform coding technique is Adaptive Transform Coding (ATC), which uses a fast transformation (such as the discrete cosine transformation) to split blocks of the speech signal into a large numbers of frequency bands. The number of bits used to code each transformation coefficient is adapted depending on the spectral properties of the speech, and toll quality reproduced speech can be achieved at bit rates as low as 16 kbits/s.



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