Speech Coding: Fundamentals and Applications

Speech coding is the process of obtaining a compact representation of voice signals for efficient transmission over band-limited wired and wireless channels and/or storage. Today, speech coders have become essential components in telecommunications and in the multimedia infrastructure. Commercial systems that rely on efficient speech coding include cellular communication, voice over internet protocol (VOIP), videoconferencing, electronic toys, archiving, and digital simultaneous voice and data (DSVD), as well as numerous PC-based games and multimedia applications. Speech coding is the art of creating a minimally redundant representation of the speech signal that can be efficiently transmitted or stored in digital media, and decoding the signal with the best possible perceptual quality. Like any other continuous-time signal, speech may be represented digitally through the processes of sampling and quantization; speech is typically quantized using either 16-bit uniform or 8-bit companded quantization. Like many other signals, however, a sampled speech signal contains a great deal of information that is either redundant (nonzero mutual information between successive samples in the signal) or perceptually irrelevant (information that is not perceived by human listeners). Most telecommunications coders are lossy, meaning that the synthesized speech is perceptually similar to the original but may be physically dissimilar. A speech coder converts a digitized speech signal into a coded representation, which is usually transmitted in frames. A speech decoder receives coded frames and synthesizes reconstructed speech. Standards typically dictate the input–output relationships of both coder and decoder. The input–output relationship is specified using a reference implementation, but novel implementations are allowed, provided that input–output equivalence is maintained. Speech coders differ primarily in bit rate (measured in bits per sample or bits per second), complexity (measured in operations per second), delay (measured in milliseconds between recording and playback), and perceptual quality of the synthesized speech. Narrowband (NB) coding refers to coding of speech signals whose bandwidth is less than 4 kHz (8 kHz sampling rate), while wideband (WB) coding refers to coding of 7-kHz-bandwidth signals (14–16 kHz sampling rate). NB coding is more common than WB coding mainly because of the narrowband nature of the wireline telephone channel (300–3600 Hz). More recently, however, there has been an increased effort in wideband speech coding because of several applications such as videoconferencing. There are different types of speech coders. Table 1 summarizes the bit rates, algorithmic complexity, and standardized applications of the four general classes of coders described in this article; Table 2 lists a selection of specific speech coding standards. Waveform coders attempt to code the exact shape of the speech signal waveform, without considering the nature of human speech production and speech perception. These coders are high-bit-rate coders (typically above 16 kbps). Linear prediction coders (LPCs), on the other hand, assume that the speech signal is the output of a linear time-invariant (LTI) model of speech production. The transfer function of that model is assumed to be all-pole (autoregressive model). The excitation function is a quasiperiodic signal constructed from discrete pulses (1–8 per pitch period), pseudorandom noise, or some combination of the two. If the excitation is generated only at the receiver, based on a transmitted pitch period and voicing information, then the system is designated as an LPC vocoder. LPC vocoders that provide extra information about the spectral shape of the excitation have been adopted as coder standards between 2.0 and 4.8 kbps. LPC-based analysis-by-synthesis coders (LPC-AS), on the other hand, choose an excitation function by explicitly testing a large set of candidate excitations and choosing the best. LPC-AS coders are used in most standards between 4.8 and 16 kbps. Subband coders are frequency-domain coders that attempt to parameterize the speech signal in terms of spectral properties in different frequency bands. These coders are less widely used than LPC-based coders but have the advantage of being scalable