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Session F - Posters 2 Friday, November 30 3:30 pm-5:00 pm


Signal Processing


3:30 pm

F-1 Hearing Aid Distortion Measurement Using the Auditory Distance Parameter (poster)

Vijay Parsa and Donald G. Jamieson, University of Western Ontario, London, Ontario, Canada

Distortion in hearing aids degrades the sound quality and reduces user satisfaction with these devices. In this paper, a distortion measure derived from the hearing aid response to natural speech stimuli is presented. The hearing aid response was modelled using a time-varying ARMA system whose coefficients were estimated using the Multiple Model Least Squares (MMLS) algorithm. The amount of distortion in the hearing aid was quantified using an "auditory distance" parameter, which computes the distance between the hearing aid response and the model output. It is shown that the auditory parameter correlates better with perceptual judgements of hearing aid sound quality, both by normal and hearing impaired listeners, than the conventional hearing aid distortion measures.

Convention Paper 5434


3:30 pm

F-2 High Performance PWM Power Audio Amplifier (poster)

Johann Gaboriau, Xiaofan Fei, and Eric Walburger, Cirrus Logic Inc., Austin, TX, USA

A method used to remove the distortion in a digital PWM amplifier is introduced. This method is based on correction factors added to each integrator of a multi-bits delta-sigma modulator loop. This results in a tremendous improvement in the distortion performance of the system. A dynamic range of 100 dB is obtained, with all harmonics supressed below 102dB. It is particularly useful for digital audio amplifiers.

Convention Paper 5428


Coding of Audio Signals


3:30 pm

F-3 High-Quality Encoding Algorithms in MPEG-2 AAC for Electric Music Distribution (poster)

Takashi Katayama, Kosuke Nishio and Masaharu Matsumoto, Matsushita Electric Industrial Co., Ltd., Kadoma, Osaka, Japan
Yoshiaki Takagi, Yasuhito Watanabe and Kazuhiro Iida, Matsushita Communication Industrial Co., Ltd., Yokohama, Kanagawa, Japan

We have developed the MPEG-2 AAC (LC profile) Encoder in the professional authoring systems for Electric music distribution (hereafter EMD). The encoders for professional use are required higher quality sounds to express music contents correctly than the encoder for consumer use. So we have researched a few quantization methods in MPEG-2 AAC encoding algorithm that we think it effect the sound quality seriously and we have developed new methods to get the high quality sounds. In this paper, we described the new methods and the construction of the encoder system for the EMD using them.

Convention Paper 5433


3:30 pm

F-4 Schemes for Re-Compressing MP3 Audio Bitstreams (poster)

Ye Wang, Juha Ojanperä, Miikka Vilermo and Mauri Väänänen, Nokia Research Center, Tampere, Finland

This paper presents three schemes for re-compressing MP3 (MPEG-1 Layer III) audio bitstreams. The first two schemes are lossless ones, which exploit the inter-frame redundancies of the main data (the scale factors and the quantized MDCT coefficients) of the MP3 bitstream. The third scheme is a lossy approach, which exploit the redundancies between consecutive beat-patterns. The aim is to study the potential of the new coding schemes. Preliminary results are demonstrated in this paper.

Convention Paper 5435


3:30 pm

F-5 Selecting an Optimal Huffman Decoder for AAC (poster)

Vladimir Z. Mesarovic, Raghunath Rao, Miroslav V. Dokic and Sachin Deo, Cirrus Logic Inc., Austin, TX, USA

In today's competitive consumer audio market the Advanced Audio Coding (AAC) format has quickly become a must-have technology with its adoption on the Internet, in digital radio, digital television and home theatre. Compression using AAC retains high audio quality even at low bit rates. One reason for this effectiveness is the use of Huffman variable length coding to represent frequency domain information. However, this requires to perform the relatively complex task of Huffman decoding in the audio decoder, which is typically very sensitive to cost and processor speed requirements. Furthermore, encoders can sometimes create worst-case scenarios consisting of very long code words, even when unnecessary. Thus, one needs to optimize the Huffman decoding for these worst-case scenarios without giving up average performance. This paper discusses various methods for Huffman decoding, their inherent implementation tradeoffs on a DSP platform and proposes improvements that are specific to the Huffman codebooks used in AAC.

Convention Paper 5436


3:30 pm

F-6 Audio Coding Using Perceptually Controlled Bitstream Buffering (poster)

Christof Faller, Agere Systems, Murray Hill, NJ, USA

Perceptual audio coders use a varying number of bits to encode subsequent frames according to the perceptual entropy of the audio signal. For transmission over a constant bitrate channel the bitstream must be buffered. The buffer must be large enough to absorb variations in the bitrate, otherwise the quality of the audio will be compromised. We present a new scheme for buffer control of perceptual audio coders. In contrast to conventional schemes the proposed scheme systematically reduces the variation in a perceptual distortion measure over time. The new scheme applied to a perceptual audio coder (PAC) improves the quality of the encoded signal for a given buffer size. The same technique can be used to increase the performance of other coders such as MPEG-1 Layer III or MPEG-2 AAC while maintaining backward compatibility.

Convention Paper 5437

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