In the past few years, objective quality assessment models have become increasingly used for assessing or monitoring speech and audio quality. By measuring perceived quality on an easily-understood subjective scale, such as listening quality (excellent, good, fair, poor, bad), these methods provide a quick and repeatable way to estimate customer experience. Typical applications include audio quality evaluation, selection of codecs or other equipment, and measuring the quality of telephone networks. To introduce this special issue, this paper provides an overview of the field, outlining the main approaches to intrusive, nonintrusive and parametric models and discussing some of their limitations and areas of future work

The move to digital is a natural progression taking place in all aspects of broadcast media applications from document processing in newspapers to video processing in television distribution. This is no less true for audio broadcasting which has taken a unique development path in the United States. This path has been heavily influenced by a combination of regulatory and migratory requirements specific to the U.S. market. In addition, competition between proposed terrestrial and satellite systems combined with increasing consumer expectations have set ambitious, and often changing, requirements for the systems. The result has been a unique set of evolving requirements on source coding, channel coding, and modulation technologies to make these systems a reality. This paper outlines the technical development of the terrestrial wireless and satellite audio broadcasting systems in the U.S., providing details on specific source and channel coding designs and adding perspective on why specific designs were selected in the final systems. These systems are also compared to other systems such as Eureka-147, DRM, and Worldspace, developed under different requirements.

This paper describes the 8 kb/s speech coding algorithm G.729 which has been standardized by ITU-T. The algorithm is based on a conjugate-structure algebraic CELP (CS-ACELP) coding technique and uses 10 ms speech frames. The codec delivers toll-quality speech (equivalent to 32 kb/s ADPCM) for most operating conditions. This paper describes the coder structure in detail and discusses the reasons behind certain design choices. A 16-b fixed-point version has been developed as part of Recommendation G.729 and a summary of the subjective test results based on a real-time implementation of this version are presented.

With the deployment of digital cellular systems such as 13 kb/s GSM RPE-LTP, 8 kb/s fixed rate IS54 VSELP and 8 kb/s variable rate IS96 QCELP it is becoming clear that these coders are not as robust as originally expected. As a result there has been renewed interest in higher bit rate coders that provide toll quality performance with regard to various input signals and multiple encoding. This paper describes the implementation of a toll quality 13 kb/s CELP coder. This coder can also operate at a variable bit rate mode (13, 6.5 and 0.8 kb/s), and has potential applications for a revised CDMA cellular system and PCS systems based on GSM.

The pitch-predictor contributes greatly to the efficiency of current analysis-by-synthesis speech coders by mapping the past reconstructed signal into the present. However, for good performance, it is required that its parameters are updated often (one every 2.5-7.5 ms). A slower update rate of the pitch-predictor delay results in time misalignment between the original signal and the pitch-predictor contribution to the reconstructed signal and the pitch-predictor contribution to the reconstructed signal. The authors introduce a new procedure, that allows a slow update rate of the pitch-predictor parameters without this problem. In this method the original signal is modified in a closed-loop fashion such that the parameter values obtained by interpolation of open-loop estimates form the optimal encoding of the modified signal. This new paradigm is a generalization of the familiar analysis-by-synthesis principle. The generalized analysis-by-synthesis principle can be used for interpolation of both the pitch-predictor delay and gain. The authors compare, by means of a subjective test, speech signals encoded with different versions of the code-excited linear predictor delay and gain. They compare, by means of a subjective test, speech signals encoded with different versions of the code-excited linear predictor (CELP) coder. The comparison shows that a pitch predictor exploiting the present interpolation strategy, with an update rate of 50 Hz, provides a subjective speed quality similar to a conventional pitch predictor where the parameters are updated for every pitch cycle.

Many modifications can be applied to a speech signal without changing its perceptual quality. For a particular speech coder, the coding efficiency will differ for distinct modifications. To exploit this, the authors introduced a generalized analysis-by-synthesis procedure. In this procedure, a search is performed over a multitude of modified original signals (on a blockwise basis), and the signal which can be encoded with the least distortion is selected for transmission. At the receiver, a quantized version of this modified original signal is constructed. The authors discuss the application of generalized analysis-by-synthesis coding to the pitch predictor of a code excited linear predictor (CELP) coder. The use of this technique makes it possible to transmit the pitch predictor parameters at a much lower rate than conventional approaches, without compromising speech quality.>

The design and implementation of a real-time CELP coder for mobile communication applications are discussed. To realize a single-chip implementation, several tradeoffs were made without compromising speech quality. In addition, techniques that make the coder more robust under a variety of channel conditions are discussed. The real-time coder can be operated at different bit rates (8, 6.8, 4.6 kb/s) by simply changing the frame update rates. The speech quality was evaluated through a formal listening test, and it was found that this coder compares favorably with other (standardized) coders operating at similar or higher rates.>

A first-order pitch predictor is described whose delay is specified as an integer number of samples plus a fraction of a sample at the current sampling rate. This realization has a better performance than conventional multiple coefficient predictors and leads to more efficient coding of the predictor parameters. Also discussed is the application of noninteger delay pitch predictors to low-bit-rate speech coding.>

The authors present results on the comparative performance of nonuniform scalar quantizers using three different LPC (linear predictive coding) representations: the arcsine of reflection coefficients, the log area ratios, and the line spectral frequencies. On comparing the spectral distortion introduced by quantizers based on these representations, it was found that the average distortion was very similar for all three, with the arcsine showing fewer large spectral errors. In a parallel study, the performance of the above LPC representations and the autocorrelation coefficients for interpolating the spectrum between adjacent time frames was investigated and revealed only small differences between the different representations. Informal listening tests with a complete 8 kb/s code-excited linear predictive (CELP) coder, incorporating both quantization and interpolation, showed no significant differences between the various LPC representations, suggesting that the random codebook for the excitation is able to compensate for small spectral deviations.>

The authors examine the robustness of the code-excited linear predictive (CELP) coder, such as its ability to cope with nonspeech and corrupted speech inputs or to survive errors in the transmission of the coder parameters. They describe how they determined the error sensitivity of each coder parameter and identified the error propagation mechanisms. They find that the coder is robust to many kinds of input signals but is sensitive to channel errors. They show that the effect of bit errors can be reduced significantly with relatively simple measures such as repetition of the coder parameters from the most recent error-free frame. Incorporating these techniques into CELP results in a robust system with an acceptable performance for both burst and random bit errors rates of up to 1%. The clear channel performance degrades slightly (0.2 dB) as a result of these techniques.>

Some of the distortions produced by CELP (code-excited linear prediction) coders are characterized. It is found that the coder does not reproduce high frequencies well and that rapid changes in the speech signal are not adequately tracked. Within the framework of the current CELP concept, strategies are discussed that can reduce these distortions. Nonstationarities in the speech signal can be better followed by allowing a flexible allocation of the bits used for the excitation. However, the bit allocation procedures and the way the bits are used need further improvement. The reproduction of higher frequencies can be improved by changing the error-weighting procedure or by shaping the code-book excitation functions.>

The general structure of this class of coders is reviewed, and the particulars of its members are discussed. The different analysis procedures are described, and the contributions of the various coder parameters to the performance of the coder are examined. Quantization procedures for each transmitted parameter are given along with examples of bit allocations. The speech quality produced by these coders is high at 16 kb/s and good at 8 kb/s, but only fair at 4.8 kb/s. The use of postprocessing techniques changes the performance at lower rates, but more research is needed to further improve the coders.>

Past research on CELP (Code-Excited Linear Predictive) coders has mainly concentrated on the feasibility of the CELP concept and on the reduction of the computational complexity. In this paper we address the problem of finding and encoding the excitation parameters with a limited bit rate, such that high quality speech coding in the 4.8 - 7.2 kb/s range becomes feasible. First, we examine the effect of the various excitation parameters such as code book size, code book population, order of the long-term predictor and update rate on the quality of the reconstructed speech. Second, we investigate procedures for designing and incorporating quantizers for the parameters involved. Finally, using both scalar and vector quantization techniques for the LPC coefficients, we simulated 4.8 kb/s and 7.2 kb/s coders. We also report on the use of postfiltering to further improve the performance of the CELP coder.

An attractive algorithm for the encoding of toll-quality speech below 16 kb/s is based on the so-called regular.pulse encoding (RPE) modeling scheme [1]. Basically. the RPE speech coder models the speech signal as the output of a linear predictive spectral envelope filter whose input is a short-time base-band approximation of the prediction residual. Although the complexity of the conceptual RPE algorithm is only moderate, we will demonsrate in this paper that under certain simplifying, albeit realistic, assumptions it turns out to be possible to come up with a scheme that combines the computational simplicity of a well designed base-band coder [2], with the subjective optimality of the uncompromisod RPE concept. Moreover, we shall also show that it is possible to overcome the problem of large encoding delay, wMch is of major concern in certain real-time applications.

This paper describes an effective and efficient time domain speech encoding technique that has an appealingly low complexity, and produces (near) toll quality speech at rates below 16 kbit/s. The proposed coder uses linear predictive techniques to remove the short-time correlation in the speech signal. The remaining (residual) information is then modeled by a regular (in time) excitation signal that, when inputted to the time-varying model filter, produces a signal that is "close" to the reference speech signal. The procedure for finding the appropriate excitation model parameters incorporates the solution of a few sets of linear equations and is of moderate complexity compared to competing coding systems such as Adaptive Transform Coding and Multi-Pulse Excitation Coding.

We report on the results obtained from simulations of the Multi-Pulse Excitation Coder as proposed by Atal and Remde [4]. We investigated the effects of the different analysis parameters on the resulting synthetic speech signals, using objective and subjective tests. We compared the in [4] proposed sub-optimal solutions with another sub-optimal solution based on an orthogonalization of the solution space and found that the original proposed solutions are reasonable choices. We demonstrate that the inclusion of a pitch detector significantly improves the perceived quality of the synthetic speech. We also describe a modification of the original algorithm, resulting in a lower complexity, and a speech quality close to the results obtained with the original algorithm.