Issue 110, September 1999
Talking Back: Adding Speech to Embedded Applications

WHAT IS ADPCM?

ADPCM is a waveform coding technique that attempts to code signals without any knowledge about how the signal was created. This implies that a waveform coder can be applied to other forms of data besides speech (e.g., video). In general, these coders are simple, with bit rates above 16 kbps. Anything lower degrades the reconstructed speech.

ADPCM is based on two principles of speech. Because there is a high correlation between consecutive speech samples, a relatively simple algorithm could be used to predict what the next sample might be, based on previous samples.

When the predicted sample was compared to the real sample, it was found that the resulting error signal had a lower variance than the original speech samples and could therefore be quantized with fewer bits. It was also found that no side information about the predictor would have to be sent if the prediction was based on the quantized samples rather than on the incoming speech signal.

The result was differential pulse code modulation, formerly named ITU-T G.721. Further studies showed that if the predictor and quantizer were made to be adaptive (i.e., that smaller samples are quantized using smaller steps and larger samples with larger steps), then the reconstructed speech more closely matched the original speech.

This adaptation helps the speech processor handle changes in the incoming speech signal more effectively. Thus the creation of ADPCM standardized to be ITU-T G.726 and G.727. Figure 2 shows a block diagram of the encoder and decoder portions of ADPCM. Notice that both the encoder and decoder share the same quantizer and predictor.

Figure 2—Because the decoder block is embedded in the encoder, the ADPCM algoritm does not need to send or store any additional side information with the compressed data.

Most DSP manufacturers can show some type of speech algorithm that has been implemented for their architecture. Very few 8-bit microcontroller manufacturers can say the same due to the horsepower required to implement the speech coding algorithms.

The ADPCM algorithm discussed in this article was developed by the now defunct Interactive Multimedia Association (IMA) based on an Intel DVI variation of the standard G.726. Normally, this algorithm is quite rigorous in the computation category, but the IMA version reduces the floating-point math and complex mathematical functions to simple arithmetic and table lookups.

A 16-bit 2’s complement speech sample is converted into a 4-bit ADPCM code. The algorithm uses approximately 600 words of program memory and 13 bytes of data memory. Almost any 8-bit microcontroller can implement this algorithm thanks to the small amount of resources required.

The source code gives the complete ADPCM encode and decode routines written for use in Microchip’s assembler (MPASM). The missing piece to the source code is that for each message recorded or played, all the registers (PrevSampleL, PrevSampleH, and PrevIndex) must be cleared.