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MC145540

ADPCM Codec

Motorola => Freescale 

In a sampling environment, Nyquist theory says that to properly sample a continuous signal, it must be

sampled at a frequency higher than twice the signal’s highest frequency component. Voice contains

spectral energy above 3 kHz, but its absence is not detrimental to intelligibility. To reduce the digital

data rate, which is proportional to the sampling rate, a sample rate of 8 kHz was adopted, consistent

with a bandwidth of 3 kHz. This sampling requires a low-pass filter to limit the high frequency energy

above 3 kHz from distorting the inband signal. The telephone line is also subject to 50/60 Hz power line

coupling, which must be attenuated from the signal by a high-pass filter before the analog-to-digital

converter.

The digital-to-analog conversion process reconstructs a staircase version of the desired inband signal

which has spectral images of the inband signal modulated about the sample frequency and its harmon-

ics. These spectral images are called aliasing components which need to be attenuated to obtain the

desired signal. The low-pass filter used to attenuate these aliasing components is typically called a

reconstruction or smoothing filter.

The MC145540 ADPCM Codec incorporates this codec function as one of its main functional blocks.

2.1.2 ADPCM Transcoder Block Description

An Adaptive Differential PCM (ADPCM) transcoder is used to reduce the data rate required to transmit a

PCM encoded voice signal while maintaining the voice fidelity and intelligibility of the PCM signal.

The ADPCM transcoder is used on both Mu-Law and A-Law 64 kbps data streams which represent

either voice or voice band data signals that have been digitized by a PCM codec-filter. The PCM to

ADPCM encoder section of this transcoder has a type of linear predicting digital filter which is trying to

predict the next PCM sample based on the previous history of the PCM samples. The ADPCM to PCM

decoder section implements an identical linear predicting digital filter. The error or difference between

the predicted and the true PCM input value is the information that is sent from the encoder to the decod-

er as an ADPCM word. The characteristics of this ADPCM word include the number of quantized steps

(this determines the number of bits per ADPCM word) and the actual meaning of this word is a function

of the predictor’s output value, the error signal, and the statistics of the history of PCM words. The term

“adaptive” applies to the transfer function of the filter that generates the ADPCM word which adapts to

the statistics of the signals presented to it. This means that an ADPCM word ‘3’ does not have the same

absolute error voltage weighting for the analog signal when the channel is quiet as it does when the

channel is processing a speech signal. The ADPCM to PCM decoder section has a reciprocating filter

function which interprets the ADPCM word for proper reconstruction of the PCM sample.

The adaptive characteristics of the ADPCM algorithm make it difficult to analyze and quantify the

v performance of the ADPCM code sequence. The 32 kbps algorithm was optimized for both voice and

moderate speed modems ( 4800 baud). This optimization includes that the algorithm supports the

voice frequency band of 300 Hz to 3400 Hz with minimal degradation for signal-to-distortion, gain-

versus-level, idle channel noise and other analog transmission performance. This algorithm has also

been subjected to audibility testing with many languages for Mean Opinion Score (MOS) ratings and

performed well when compared to 64 kbps PCM. The standards committees have specified multiple

16000 word test vectors for the encoder and for the decoder to verify compliance. To run these test

vectors, the device must be initialized to the reference state by resetting the device.

In contrast to 64 kbps PCM, the ADPCM words appear as random bit activity on an oscilloscope display

whether the audio channel is processing speech or a typical PCM idle channel with nominal bit activity.

The ADPCM algorithm does not support dc signals with the exception of digital quiet, which will result in

all ones in the ADPCM channel. All digital processing is performed on 13-bit linearizations of the 8-bit

PCM companded words, whether the words are Mu-Law or A-Law. This allows an ADPCM channel to

be intelligibly decoded into a Mu-Law PCM sequence or an A-Law PCM sequence irrespective of

whether it was originally digitized as Mu-Law or A-Law. There will be additional quantizing degradation if

the companding scheme is changed because the ADPCM algorithm is trying to reconstruct the original

13-bit linear codes, which included companding quantization.

MOTOROLA

MC145540

2-3

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