Peter Noll

Adaptive transform coding of speech signals

This paper discusses speech coding systems based upon transform coding (TC). It compares several transforms and shows that the cosine transform leads to a nearly optimum performance for almost all speech sounds. Various adaptive coding strategies are then investigated, and a coding scheme is proposed that is based on a nonadaptive discrete cosine transform (DCT), on an adaptive bit assignment, and on adaptive quantization. The adaptation is controlled by a short-term basis spectrum that is derived from the transform coefficients prior to coding and transmission and that is transmitted as side information to the receiver. The main result is that this adaptive transform coder performs better than all known nonpitch-tracking coding schemes; it extends the range of speech waveform coding to lower bit rates and closes the gap between vocoders and predictive waveform coders.

MPEG digital audio coding

The Moving Pictures Expert Group (MPEG) within the International Organization of Standardization (ISO) has developed a series of audio-visual standards known as MFEG-1 and MPEG-2. These audio-coding standards are the first international standards in the field of high-quality digital audio compression. MPEG-1 covers coding of stereophonic audio signals at high sampling rates aiming at transparent quality, whereas MPEG-2 also offers stereophonic audio coding at lower sampling rates. In addition, MPEG-2 introduces multichannel coding with and without backwards compatibility to MPEG-1 to provide an improved acoustical image for audio-only applications and for enhanced television and video-conferencing systems. MPEG-2 audio coding without backwards compatibility, called IMPEG-2 Advanced Audio Coding (AAC), offers the highest compression rates. Typical application areas for MPEG-based digital audio are in the fields of audio production, program distribution and exchange, digital sound broadcasting, digital storage, and various multimedia applications. We describe in some detail the key technologies and main features of MPEG-1 and MPEG-2 audio coders. We also present the MPEG-4 standard and discuss some of the typical applications for MPEG audio compression.

Wideband speech and audio coding

Typical parameters of wideband speech and audio signals, including digitized versions of each, potential applications, and available transmission media, are described. Facts about human auditory perception that are exploited in audio coding and quality measures that play an important role in coder evaluations and designs are reviewed. Techniques for efficient coding of wideband speech and audio signals, with an emphasis on existing standards, are discussed. The audio coding standard developed by the Moving Pictures Expert Group within the International Organization for standardization (ISO/MPEG) is covered in some detail, since it will be used in many application areas, including digital storage, transmission, and broadcasting of audio-only signals and audiovisual applications such as videotelephony, videoconferencing, and TV broadcasting. Ongoing research and standardization work is outlined.<<ETX>>

A study of complexity and quality of speech waveform coders

This paper presents the results of a pilot study comparing four different speech waveform coding techniques of varying complexity. Coder transmission rates of 24, 16, and 9.6 Kb/s were used in the experiment. Subjective ratings and objective measurements of quality are obtained and compared. A number of conclusions are drawn concerning the quality and complexity, of different coding techniques. By comparing the objective measurements to the subjective ratings a number of conclusions are also drawn concerning the strengths and weaknesses of various (objective) quality measures of speech waveform coders.

Digital audio coding for visual communications

Current and future visual communications for applications such as broadcasting videotelephony, video- and audiographic-conferencing, and interactive multimedia services assume a substantial audio component. Even text, graphics, fax, still images, email documents, etc. will gain from voice annotation and audio clips. A wide range of speech, wideband speech, and wideband audio coders is available for such applications. In the context of audiovisual communications, the quality of telephone-bandwidth speech is acceptable for some videotelephony and videoconferencing services. Higher bandwidths (wideband speech) may be necessary to improve the intelligibility and naturalness of speech. High quality audio coding including multichannel audio will be necessary in advanced digital TV and multimedia services. This paper explains basic approaches to speech, wideband speech, and audio bit rate compressions in audiovisual communications. These signal classes differ in bandwidth, dynamic range, and in listener expectation of offered quality. It will become obvious that the use of our knowledge of auditory perception helps minimizing perception of coding artifacts and leads to efficient low bit rate coding algorithms which can achieve substantially more compression than was thought possible only a few years ago. The paper concentrates on worldwide source coding standards beneficial for consumers, service providers, and manufacturers. >

Bounds on Quantizer Performance in the Low Bit-Rate Region

This paper investigates the optimum entropy versus distortion performance of quantizers optimized for uniform, Gaussian, Laplacian, and gamma-distributed memoryless sources which are useful models of the quantizer input signals in speech or picture coding schemes. We list the maximally obtainable signal-to-quantization noise ratios for one-dimensional optimum (i.e. entropy-coded) quantizers in the important low bit-rate region. These results have been obtained by an iterative solution of a set of nonlinear equations. Additionally we have also computed the corresponding rate-distortion functions by employing the Blahut-algorithm. These latter results upperbound the performances of multi-dimensional quantization schemes, and a comparison with the former results indicates the penalty to be paid for restricting a coder to perform a one-dimensionai quantization. It will be shown that the differences can be significant in the low bit-rate region.

A new image coding technique unifying fractal and transform coding

We present a new image coding scheme based on an unification of fractal and transform coding. We introduce a generalization of the luminance transformation generally used by fractal coding schemes. By extending the luminance transformation to the frequency domain fractal and transform coding become subsets of the proposed transformation. Our new coding scheme FTC (fractal based transform coding) combines the advantages of both techniques. Compared to JPEG a coding gain of 1.5-2.5 dB (PSNR) is obtained. The encoding time is reduced compared to conventional fractal coding schemes and a better convergence at the decoder is attained. At equal error rates the subjective quality of images coded with the new scheme is superior compared to transform coded images.<<ETX>>

Approaches to adaptive transform speech coding at low bit rates

This paper discusses problems of adaptive transform coding schemes at bit rates of 12 kbit/s and below. Objective and subjective performance reductions, like low-pass filtering effects as one of the main sources of perceptual distortion, are investigated and proposals are made how to improve the performance of the coder at low and medium bit rates. Additionally, the needed transmission of side information reduces the efficiency of the scheme. Various methods to lower the rate of this supplementary data signal are given as well as modifications of the scheme which lead to a more easily implemented coder structure.

Audio coding with a dynamic wavelet packet decomposition based on frequency-varying modulated lapped transforms

Optimum time-frequency decompositions are very useful in audio coding applications, because the signal energy can be maximally concentrated even for the wide variety of audio signal characteristics. Moreover, this signal representation is particularly well suited for a perceptual weighting of the quantization noise. The well known tree structure of cascaded 2-channel filterbanks allows a very flexible optimization, leading to a signal adaptive, dynamic wavelet packet decomposition. A major drawback of this technique are strong spectral side lobes which produce clearly audible aliasing in perceptual coders. We present a new dynamic wavelet packet decomposition, based on modulated lapped transforms, which allows the same flexibility while avoiding the disadvantage mentioned above. We propose a scheme for low bit rate audio coding that efficiently exploits the high energy concentration. This new codec yields excellent audio quality at about 55 kb/s for monophonic signals.

Multipath Search Coding of Stationary Signals with Applications to Speech

This paper deals with the application of multipath search coding (MSC) concepts to the coding of stationary memoryless and correlated sources and of speech signals at a rate of one bit per sample. We have made use of three MSC classes: 1) codebook coding (vector quantization), 2) tree coding, and 3) trellis coding. This paper explains the performances of these coders and compares them both with those of conventional coders and with rate-distortion bounds. Figs. 2 and 3 demonstrate the potentials of MSC coding strategies. The paper reports also on results of MSC coding of speech, where both the strategy of adaptive quantization and of adaptive prediction were included in coder design.