Time and frequency domain algorithms for speech coding

The promise of digital hardware economies (due to recent advances in VLSI technology), has focussed much attention on more complex and sophisticated speech coding algorithms which offer improved quality at relatively low bit rates. This thesis describes the results (obtained from computer simulations) of research into various efficient (time and frequency domain) speech encoders operating at a transmission bit rate of 16 Kbps. In the time domain, Adaptive Differential Pulse Code Modulation (ADPCM) systems employing both forward and backward adaptive prediction were examined. A number of algorithms were proposed and evaluated, including several variants of the Stochastic Approximation Predictor (SAP). A Backward Block Adaptive (BBA) predictor was also developed and found to outperform the conventional stochastic methods, even though its complexity in terms of signal processing requirements is lower. A simplified Adaptive Predictive Coder (APC) employing a single tap pitch predictor considered next provided a slight improvement in performance over ADPCM, but with rather greater complexity. The ultimate test of any speech coding system is the perceptual performance of the received speech. Recent research has indicated that this may be enhanced by suitable control of the noise spectrum according to the theory of auditory masking. Various noise shaping ADPCM configurations were examined, and it was demonstrated that a proposed pre-/post-filtering arrangement which exploits advantageously the predictor-quantizer interaction, leads to the best subjective performance in both forward and backward prediction systems. Adaptive quantization is instrumental to the performance of ADPCM systems. Both the forward adaptive quantizer (AQF) and the backward oneword memory adaptation (AQJ) were examined. In addition, a novel method of decreasing quantization noise in ADPCM-AQJ coders, which involves the application of correction to the decoded speech samples, provided reduced output noise across the spectrum, with considerable high frequency noise suppression. More powerful (and inevitably more complex) frequency domain speech coders such as the Adaptive Transform Coder (ATC) and the Sub-band Coder (SBC) offer good quality speech at 16 Kbps. To reduce complexity and coding delay, whilst retaining the advantage of sub-band coding, a novel transform based split-band coder (TSBC) was developed and found to compare closely in performance with the SBC. To prevent the heavy side information requirement associated with a large number of bands in split-band coding schemes from impairing coding accuracy, without forgoing the efficiency provided by adaptive bit allocation, a method employing AQJs to code the sub-band signals together with vector quantization of the bit allocation patterns was also proposed. Finally, 'pipeline' methods of bit allocation and step size estimation (using the Fast Fourier Transform (FFT) on the input signal) were examined. Such methods, although less accurate, are nevertheless useful in limiting coding delay associated with SRC schemes employing Quadrature Mirror Filters (QMF)

Quality aspects of Internet telephony

Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication. The first four chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research. The appendix of this dissertation constitutes the research contributions. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to handover calls from 802.11 networks to the cellular infrastructure. We then consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice stream given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process. The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today

Perceptual models in speech quality assessment and coding

The ever-increasing demand for good communications/toll quality speech has created a renewed interest into the perceptual impact of rate compression. Two general areas are investigated in this work, namely speech quality assessment and speech coding. In the field of speech quality assessment, a model is developed which simulates the processing stages of the peripheral auditory system. At the output of the model a "running" auditory spectrum is obtained. This represents the auditory (spectral) equivalent of any acoustic sound such as speech. Auditory spectra from coded speech segments serve as inputs to a second model. This model simulates the information centre in the brain which performs the speech quality assessment. [Continues.

High-fidelity imaging : the computational models of the human visual system in high dynamic range video compression, visible difference prediction and image processing

As new displays and cameras offer enhanced color capabilities, there is a need to extend the precision of digital content. High Dynamic Range (HDR) imaging encodes images and video with higher than normal bit-depth precision, enabling representation of the complete color gamut and the full visible range of luminance. This thesis addresses three problems of HDR imaging: the measurement of visible distortions in HDR images, lossy compression for HDR video, and artifact-free image processing. To measure distortions in HDR images, we develop a visual difference predictor for HDR images that is based on a computational model of the human visual system. To address the problem of HDR image encoding and compression, we derive a perceptually motivated color space for HDR pixels that can efficiently encode all perceivable colors and distinguishable shades of brightness. We use the derived color space to extend the MPEG-4 video compression standard for encoding HDR movie sequences. We also propose a backward-compatible HDR MPEG compression algorithm that encodes both a low-dynamic range and an HDR video sequence into a single MPEG stream. Finally, we propose a framework for image processing in the contrast domain. The framework transforms an image into multi-resolution physical contrast images (maps), which are then rescaled in just-noticeable-difference (JND) units. The application of the framework is demonstrated with a contrast-enhancing tone mapping and a color to gray conversion that preserves color saliency.Aktuelle Innovationen in der Farbverarbeitung bei Bildschirmen und Kameras erzwingen eine Präzisionserweiterung bei digitalen Medien. High Dynamic Range (HDR) kodieren Bilder und Video mit einer grösseren Bittiefe pro Pixel, und ermöglichen damit die Darstellung des kompletten Farbraums und aller sichtbaren Helligkeitswerte. Diese Arbeit konzentriert sich auf drei Probleme in der HDR-Verarbeitung: Messung von für den Menschen störenden Fehlern in HDR-Bildern, verlustbehaftete Kompression von HDR-Video, und visuell verlustfreie HDR-Bildverarbeitung. Die Messung von HDR-Bildfehlern geschieht mittels einer Vorhersage von sichtbaren Unterschieden zweier HDR-Bilder. Die Vorhersage basiert dabei auf einer Modellierung der menschlichen Sehens. Wir addressieren die Kompression und Kodierung von HDR-Bildern mit der Ableitung eines perzeptuellen Farbraums für HDR-Pixel, der alle wahrnehmbaren Farben und deren unterscheidbaren Helligkeitsnuancen effizient abbildet. Danach verwenden wir diesen Farbraum für die Erweiterung des MPEG-4 Videokompressionsstandards, welcher sich hinfort auch für die Kodierung von HDR-Videosequenzen eignet. Wir unterbreiten weiters eine rückwärts-kompatible MPEG-Kompression von HDR-Material, welche die übliche YUV-Bildsequenz zusammen mit dessen HDRVersion in einen gemeinsamen MPEG-Strom bettet. Abschliessend erklären wir unser Framework zur Bildverarbeitung in der Kontrastdomäne. Das Framework transformiert Bilder in mehrere physikalische Kontrastauflösungen, um sie danach in Einheiten von just-noticeable-difference (JND, noch erkennbarem Unterschied) zu reskalieren. Wir demonstrieren den Nutzen dieses Frameworks anhand von einem kontrastverstärkenden Tone Mapping-Verfahren und einer Graukonvertierung, die die urspr ünglichen Farbkontraste bestmöglich beibehält