When digital systems started replacing analog equipment a few decades ago, processing power was scarce and expensive, and coding techniques still primitive. For those reasons, most early digital systems used a very simple coding scheme: PCM (Pulse Code Modulation).
In this digital representation of speech, there isn't really any coding in the compression sense. The signal is simply sampled and quantized. More specifically, the speech signal is typically sampled at 8 KHz, and each sample is encoded with 8-bit precision, using one of two quantization schemes, usually referred to as A-law and µ-law. This gives a total rate of 64 Kbps.
The PCM system used in telephony has been standardized by the ITU (International Telecommunication Union) in the standard G.711 [1]. For Voice over Internet Protocol (VoIP) or other packet network applications, the speech samples will be grouped into frames (typically 10 ms in duration) and sent as packets across the network, one frame per packet. Note that a frame corresponds to a data unit in the terminology of Chapter 2. Note that, since there is no real coding, there is no dependence across packets: packets can be received and decoded independently.
When G.711 was first adopted, the main motivation was quality: A digital signal was not subject to degradation. At the same time, a 64-Kbps digital channel had a significant cost, and there was a strong push toward increased compression. With the evolution of speech compression technology, and increased processing power, more complex speech codecs were also standardized (e.g., [3"6]), providing better compression.
Curiously, today, in many applications bandwidth is not necessarily a significant constraint any more, and we are starting to see basic PCM-coded speech increasing in usage again. Furthermore, many error concealment techniques operate in the time domain, and therefore are best understood as applying to PCM-coded speech. For this reason, in this section we review the basic concept of packet loss as applied to speech and look at some common techniques to conceal loss in PCM coded speech.
We assume speech samples are PCM coded and grouped in 10-ms frames before transmission. Since we assume packets are either received error free or not received at all, this implies that any loss incurred in the transmission process will imply a missing segment of 10 ms (or a multiple thereof). Figure 3.1 shows a segment of a speech signal. The signal is typical of a voiced phoneme.
FIGURE 3.1: (a) A typical speech signal. (b) Original signal with two missing frames. (c) Concealed loss using Appendix I of G.711.
Figure 3.1(a) shows the original signal, whereas 3.1(b) shows a plot where 20 ms (i.e., two packets) is missing. As can be inferred from the picture, a good concealment algorithm would try to replace the missing segment by extending the prior signal with new periods of similar waveforms. This can be done with different levels of complexity, yielding also different levels of artifacts. We will now investigate a simple concealment technique, described in the Appendix I of Recommendation G.711 [2]. The results of applying that algorithm are illustrated in Figure 3.1(c).
