MPEG-1 Layer II offers standardized audio compression but does not provide data rates low enough to allow audio streaming over the Internet through low bandwidth modems. Methods are introduced in this project to reduce the bandwidth requirements of MPEG-1 Layer II audio data. New tools are developed to implement these methods and comparisons are made between the new very low bitrate files, Real Audio 3.0, and I-Wave. Pre and post processing methods are also discussed.

1. Introduction
2. Data Communication Networks
2.1 The Structure of the Internet
2.1.1 Local Area Networks (LANs)
2.1.2 Wide Area Networks (WANs)
2.1.3 The Public-switched Data Network
2.1.4 Dial-up Connections
2.1.5 Transmission Control Protocol / Internet Protocol (TCP/IP)
2.1.6 TCP vs. UDP data transmission
2.2 Digital Audio

2.2.1 Sample Rate
2.2.2 Quantization Level
2.2.3 Bandwidth Considerations

3. Audio Compression

3.1 Audio Compression Schemes

3.1.1 Time Domain Codecs

3.1.1.1 Speech Applications

3.1.1.1.1 Vocoders and Linear Predictive Coding (LPC)
3.1.1.1.2 Code Excited Linear Prediction Voice Coder (CELP)
3.1.1.1.3 Global System for Mobile communications (GSM)

3.1.1.2 Music Applications

3.1.1.2.1 m-law and A-law Companding
3.1.1.2.2 Adaptive Differential Pulse-Code Modulation (ADPCM)

3.1.2 Frequency Domain Codecs

3.1.2.1 MPEG

3.1.3 Encoder Selection Process

3.2 MPEG Overview
3.3 MPEG-1

3.3.1 Layer I and Layer II
3.3.2 Psychoacoustical Model 1

3.3.2.1 Step 1 - FFT Analysis
3.3.2.2 Step 2 - Determination of the sound pressure level in each subband.
3.3.2.3 Step 3 - Determination of the threshold in quiet (absolute threshold).
3.3.2.4 Step 4 - Finding of tonal and non-tonal components.
3.3.2.5 Step 5 - Decimation of tonal and non-tonal masking components.
3.3.2.6 Step 6 - Calculation of the individual masking thresholds.
3.3.2.7 Step 7 - Calculation of the global masking threshold LTg
3.3.2.8 Step 8 - Determination of the minimum masking threshold in each subband.
3.3.2.9 Step 9 - Calculation of the signal-to-mask ratio in each subband.

3.3.3 Layer III Encoding
3.3.4 Psychoacoustic Model 2
3.3.5 Layer I and II Decoding

4. Project Overview

4.1 Objective
4.2 Attempted Methods

4.2.1 Delta Coding and/or Prediction of Spectral Lines

5. Design and Implementation

5.1 Resources

5.1.1 The Fraunhofer Institute's MPEGiis encoder
5.1.2 Cool Edit
5.1.3 Maplay decoder

5.2 Software Modifications

5.2.1 Cool Edit MPEG encoder

5.2.1.1 Maximum Number of Bits
5.2.1.2 Bit Pool scaling
5.2.1.3 Set scaling and temporal control
5.2.1.4 Starting Points Drop Down List
5.2.1.5 Coolmpeg.txt output
5.2.1.6 Coolmpeg.ini

5.2.2 Cool Edit MPEG decoder

5.3 Pre-processing
5.4 Post-processing

6. Results

6.1 Qualitative tests

6.1.1 Voice
6.1.2 Pop
6.1.3 Rock
6.1.4 Rock Guitar
6.1.5 Hard Rock
6.1.6 Jazz
6.1.7 Classical

6.2 Compatibility with other decoders

6.2.1 Cool Edit / Fraunhofer MPEGiis
6.2.2 MaPlay
6.2.3 Phillips
6.2.4 Resonate

7. Conclusions
8. Directions for Further Study
9. References