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The Interactive Video Data System (IVDS) project began with an initial abstract concept of achieving interactive television by transmitting hidden digital information in the audio of commercials. Over the course of three years such a communication method was successfully developed, the hardware systems to realize the application were designed and built, and several full-scale field tests were conducted. The novel coding scheme satisfies all of the design constraints imposed by the project sponsors. By taking advantage of psychoacoustic properties, the hidden digital signature is inaudible to most human observers yet is detectable by the hardware decoder. The communication method is also robust against most extraneous room noise as well as the wow and flutter of videotape machines. The hardware systems designed for the application have been tested and work as intended. A triple-stage audio amplifier buffers the input signal, eliminates low frequency interference such as human voices, and boosts the filtered result to an appropriate level. A codec samples the filtered and amplified audio, and feeds it into the digital signal processor. The DSP, after applying a pre-emphasis and compensation filter, performs the data extraction by calculating FFTs, compensating for frequency shifts, estimating the digital signature, and verifying the result via a cyclic redundancy check. It then takes action appropriate for the command specified in the digital signature. If necessary it will verbally prompt and provide information to the user, and will decode infrared signals from a remote control. The results of interactions are transmitted by radio frequency spread spectrum to a cell cite, where they are then forwarded to the host computer. / Master of Science

Thesis (M.Phil.) - University of Queensland, 2005. / Includes bibliography.

Thesis (M.S..)--Florida State University, 2006. / Advisor: Steven McClung, Florida State University, College of Communication, Dept. of Communication. Title and description from dissertation home page (viewed June 7, 2006). Document formatted into pages; contains vi, 54 pages. Includes bibliographical references.

This thesis studies the application of the wavelet filter bank (WFB) in perceptual audio coding by providing brief overviews of perceptual coding, psychoacoustics, wavelet theory, and existing wavelet coding algorithms. Furthermore, it describes the poor frequency localization property of the WFB and explores one filter design method, in particular, for improving channel separation between the wavelet bands. A wavelet audio coder has also been developed by the author to test the new filters. Preliminary tests indicate that the new filters provide some improvement over other wavelet filters when coding audio signals that are stationary-like and contain only a few harmonic components, and similar results for other types of audio signals that contain many spectral and temporal components. It has been found that the WFB provides a flexible decomposition scheme through the choice of the tree structure and basis filter, but at the cost of poor localization properties. This flexibility can be a benefit in the context of audio coding but the poor localization properties represent a drawback. Determining ways to fully utilize this flexibility, while minimizing the effects of poor time-frequency localization, is an area that is still very much open for research.

<p>Ogg Vorbis is a rather new audio format with some similarities with other more known formats such as MP3 and WMA. It is generally accepted to have a better audio quality than most competing formats and it is in contrast to many of its competitors totally licence and royalty free. </p><p>The goal with this thesis is to port the existing fixed point decoder Tremor, which is written in C, to Motorola's DSP56002. The DSP has a very limited amount of memory so some optimizations has to be made to be able to run Tremor successfully. </p><p>The report presents the necessary steps taken to port Tremor to the DSP and the difficulties of this process. It also describes the memory and CPU usage of the DSP when running Tremor and other results of the port. </p><p>A description as well as examples and workarounds of bugs found in the compiler g56k is attached to this report.</p>

Ogg Vorbis is a rather new audio format with some similarities with other more known formats such as MP3 and WMA. It is generally accepted to have a better audio quality than most competing formats and it is in contrast to many of its competitors totally licence and royalty free. The goal with this thesis is to port the existing fixed point decoder Tremor, which is written in C, to Motorola's DSP56002. The DSP has a very limited amount of memory so some optimizations has to be made to be able to run Tremor successfully. The report presents the necessary steps taken to port Tremor to the DSP and the difficulties of this process. It also describes the memory and CPU usage of the DSP when running Tremor and other results of the port. A description as well as examples and workarounds of bugs found in the compiler g56k is attached to this report.

This thesis studies the application of the wavelet filter bank (WFB) in perceptual audio coding by providing brief overviews of perceptual coding, psychoacoustics, wavelet theory, and existing wavelet coding algorithms. Furthermore, it describes the poor frequency localization property of the WFB and explores one filter design method, in particular, for improving channel separation between the wavelet bands. A wavelet audio coder has also been developed by the author to test the new filters. Preliminary tests indicate that the new filters provide some improvement over other wavelet filters when coding audio signals that are stationary-like and contain only a few harmonic components, and similar results for other types of audio signals that contain many spectral and temporal components. It has been found that the WFB provides a flexible decomposition scheme through the choice of the tree structure and basis filter, but at the cost of poor localization properties. This flexibility can be a benefit in the context of audio coding but the poor localization properties represent a drawback. Determining ways to fully utilize this flexibility, while minimizing the effects of poor time-frequency localization, is an area that is still very much open for research.

Digital audio is increasingly becoming more and more a part of our daily lives. Unfortunately, the excessive bitrate associated with the raw digital signal makes it an extremely expensive representation. Applications such as digital audio broadcasting, high definition television, and internet audio, require high quality audio at low bitrates. The field of audio coding addresses this important issue of reducing the bitrate of digital audio, while maintaining a high perceptual quality. Developing an efficient audio coder requires a detailed analysis of the audio signals themselves. It is important to find a representation that can concisely model any general audio signal. In this thesis, we propose two new high quality audio coders based on two different audio representations - the sinusoidal-wavelet representation, and the warped linear predictive coding (WLPC)-wavelet representation. In addition to high quality coding, it is also important for audio coders to be flexible in their application. With the increasing popularity of internet audio, it is advantageous for audio coders to address issues related to real-time audio delivery. The issue of bitstream scalability has been targeted in this thesis, and therefore, a third audio coder capable of bitstream scalability is also proposed. The performance of each of the proposed coders was evaluated by comparisons with the MPEG layer III coder. The first coder proposed is based on a hybrid sinusoidal-wavelet representation. This assumes that each frame of audio can be modelled as a sum of sinusoids plus a noisy residual. The discrete wavelet transform (DWT) is used to decompose the residual into subbands that approximate the critical bands of human hearing. A perceptually derived bit allocation algorithm is then used to minimise the audible distortions introduced from quantising the DWT coefficients. Listening tests showed that the coder delivers near-transparent quality for a range of critical audio signals at G4 kbps. It also outperforms the MPEG layer III coder operating at this same bitrate. This coder, however, is only useful for high quality coding, and is difficult to scale to operate at lower rates. The second coder proposed is based on a hybrid WLPC-wavelet representation. In this approach, the spectrum of the audio signal is estimated by an all pole filter using warped linear prediction (WLP). WLP operates on a warped frequency domain, where the resolution can be adjusted to approximate that of the human auditory system. This makes the inherent noise shaping of the synthesis filter even more suited to audio coding. The excitation to this filter is transformed using the DWT and perceptually encoded. Listening tests showed that near-transparent coding is achieved at G4 kbps. The coder was also found to be slightly superior to the MPEG layer III coder operating at this same bitrate. The third proposed coder is similar to the previous WLPC-wavelet coder, but modified to achieve bitstream scalability. A noise model for high frequency components is included to keep the overall bitrate low, and a two stage quantisation scheme for the DWT coefficients is implemented. The first stage uses fixed rate scalar and vector quantisation to provide a coarse approximation of the coefficients. This allows for low bitrate, low quality versions of the input signal to be embedded in the overall bitstream. The second stage of quantisation adds detail to the coefficients, and hence, enhances the quality of the output signal. Listening tests showed that signal quality gracefully improves as the bitrate increases from 16 kbps to SO kbps. This coder has a performance that is comparable to the MPEG layer III coder operating at a similar (but fixed) bitrate.