Schuyler Quackenbush

Intellectual property protection systems and digital watermarking

Adequate protection of digital copies of multimedia content - both audio and video - is a prerequisite to the distribution of this content over networks. Until recently digital audio and video content has been protected by its size: it is difficult to distribute and store without compression. Modern compression algorithms allow substantial bitrate reduction while maintaining high-fidelity reproduction. If distribution of these algorithms is controlled, cleartext uncompressed content is still protected by its size. However, once the compression algorithms are generally available cleartext content becomes extremely vulnerable to piracy. In this paper we explore the implications of this vulnerability and discuss the use of compression and watermarking in the control of piracy.

Overview of MPEG-7 audio

MPEG-7 is a new ISO standard that facilitates searching for media content much as current text-based search engines ease retrieval of HTML content. This paper gives an overview of the MPEG-7 audio standard, in terms of the applications it might support, its structure, the process by which it was developed, and its specific descriptors and description schemes.

New directions in subband coding

Two very different subband coders are described. The first is a modified dynamic bit-allocation-subband coder (D-SBC) designed for variable rate coding situations and easily adaptable to noisy channel environments. It can operate at rates as low as 12 kb/s and still give good quality speech. The second coder is a 16-kb/s waveform coder, based on a combination of subband coding and vector quantization (VQ-SBC). The key feature of this coder is its short coding delay, which makes it suitable for real-time communication networks. The speech quality of both coders has been enhanced by adaptive postfiltering. The coders have been implemented on a single AT&T DSP32 signal processor. >

Covert audio watermarking using perceptually tuned signal independent multiband phase modulation

An algorithm for covert digital audio watermarking is presented. The method embeds a low-rate (in the neighborhood of 20–30 b/sec) watermark via perceptually insignificant long term multiband phase modulation, The watermark is very difficult to either discover or recover without the original (unmodulated) signal. The algorithm is shown to be robust to perceptual coding.

MPEG Surround

MPEGs most recent effort to progress the state of the art is the MPEG Surround work item. It provides an efficient method for coding multichannel sound via the transmission of a compressed stereophonic (or even monophonic) audio program plus a low-rate side-information channel. Benefits of this approach include backward compatibility with pervasive stereo playback systems while permitting next-generation players to reconstruct high-quality multichannel sound.

Coding of natural audio in MPEG-4

MPEG-4 standardizes natural audio coding at bit rates ranging from 2 kbit/s, suitable for intelligible speech coding, to 64 kbit/s per channel, suitable for high-quality audio coding. Within this range, three categories of coding are defined: parametric coding, code excited linear predictive coding (CELP) and time/frequency (T/F) coding. The unique contribution of MPEG-4 audio is that not only does it scale across a wide range of bit rates, but it also scales across a broad set of other parameters, such as sampling rate, bandwidth, voice pitch and complexity. This paper presents an overview of the MPEG-4 natural audio coding framework and each of its component coding techniques.

MPEG Unified Speech and Audio Coding

The MPEG Audio Subgroup has a rich history of accomplishments in creating music coding technology. At higher bit rates, MPEG technology can represent arbitrary sounds, including the human voice, with excellent quality. MPEG-1 and MPEG-2 Audio coders use perceptually shaped quantization noise as the primary tool for achieving compression. The MPEG-4 High-Efficiency Advanced Audio Coding (AAC) standard is a single technology capable of compressing speech, speech mixed with music, or music signals with quality that is always at least as good as the best of two state-of-the-art reference codecs, one optimized for speech and mixed content (AMR-WB B;) and the other optimized for music and general audio (HE-AACv2). This article provides an overview of the USAC architecture and summarizes the performance relative to the best state-of-the-art speech and audio codecs.