Dolby AC3 audio codec and MPEG-2 Advanced Audio Coding Recommended by

H.S.MAZEN

Faculty of Computer & Information Technology, Department of Software Engineering

Al-Madinah International University

dr.shadi@mediu.edu.my

Abstract: the digital audio coding will be

studied and MPEG-1 coding will be studied

as an example for this generic coding

system.

i. Introduction

AC3 was proposed and developed by Dolby

Inc. for DVD, HDTV, and home theater

systems. The AC3 system uses human

psychoacoustic features to mask the

inaudible audio signals. Figure 1. shows the

AC3 encoding procedure.

Overlapping blocks of 512 time samples are

windowed and transformed into the

frequency domain. Binary

exponent/mantissa coding is used to

represent quantized frequency coefficients.

Exponents indicate number of leading zeros

and act as scale factor for each mantissa. It

is referred to as spectral envelope.

Exponents for each channel are

differentially encoded. Mantissas are the

remaining significant figures of each

coefficient. It is quantized by spectral

envelope information. Mantissa

quantization referred to as core bit

allocation. A sophisticated psychoacoustic

bit allocation process is used in the

encoder. The AC3 syntax provides for delta

bit allocations to be sent to the decoder.

Figure 2. shows the AC3 decoder which

reverses the encoding steps and converts

the AC3 bitstream into the original time-

domain waveform.

Figure 3.14: An AC3 encoding procedure (Davidson et.

al. (1995)).

Figure 3.15: An AC3 encoding procedure (Davidson et.

al. (1995)).

ii. Bit allocation in AC3

AC3 uses a backward adaptive method. A

backward adaptive bit-allocation scheme

generates the bit-allocation information

from the coded audio data. This scheme

does not depend on any transmitted

information from the encoder. This scheme

has the advantage that none of the

available data rate is used to deliver the

allocation information to the decoder. So,

all the bits are used in coding audio.

However, backward adaptive allocation

scheme has the disadvantage that the bit

allocation must be computed in the

decoder from information contained in the

bitstream. The bit allocation has limited

accuracy and may contain small errors.

iii. Filter bank

The AC3 takes the overlapping blocks of 512

windowed samples and transforms them

into 256 frequency-domain points.

iv. MPEG-2 Advanced Audio Coding

The cooperated research and development

efforts from the audio coding laboratories,

such as Fraunhofer Institute, Dolby, Sony,

and AT&T led to the development of MPEG-

2 Advanced Audio Coding. AAC format has

the following features:

1) It can support up to 48 full-frequency

sound channels.

2) It can support 16 low frequency

enhancement channels.

3) It supports sampling rates up to 96 kHz,

twice the maximum afforded by MP3

and AC3.

AAC uses a modular approach which can be

summarized as (as shown in Figure 3.16):

(1) Filter bank: The AAC filter bank uses:

- plain MDCT together with an increased

window.

- adapted window shape function.

- transform block switching.

(2) Temporal noise shaping (TNS): Linear

prediction (LP) is applied across

frequency since for an impulsive

(transient) time signal it exhibits tonally

in the frequency domain.

(3) Prediction: A time domain prediction

module can be used to enhance the

compressibility of stationary audio.

(4) Middle/Side (MS) stereo: One can toggle

middle or side stereo on a subband

basis instead of on an entire-frame

basis.

(5) Quantization: By changing quantization

resolution, the given bitrate can be used

more efficiently.

(6) Huffman coding: One scale factor

Huffman codebook and 11 spectrum

Huffman codebooks are used.

(7) Bitstream format: Tradeoffs between

quality and complexity are achieve using

three profiles:

- main profile,

- low-complexity profile, and

- scalable sample rate profile

Figure 3.16: The modular coding structure of an MPEG-

2 AAC (Church (2006)).

v. Summary

In this topic the Dolby AC3 audio codec and

MPEG-2 Advanced Audio Coding (AAC) are

studied as advanced audio coding systems.

References

1. Steve Church (2006), “On beer and

audio coding; why something called AAC

is cooler than a fine Pilsner, and how it

got to be that way,” Tech. Talk of Telos

Systems,

http://www.telossystems.com/techtalk/a

acpaper_2/AAC_3.pdf.

2. G. A. Davidson, M. A. Isnardi, L. D.

Fielder, M. S. Goldman, and C. C. Todd

(2006), “ATSC video and audio coding,”

Proc IEEE, 94(1): pp. 60–76.

3. Jenq-Neng Hwang (2009),

“Multimedia Networking: From Theory to

Practice”. Cambridge University Press.

4. S. Meltzer and G. Moser (2006),

“MPEG-4 HE-AAC v2: audio coding for

today’s digital media world,” EBU

Technical Review.

5. T. Painter and A. Spanias (2004),

“Perceptual coding of digital audio,” Proc.

IEEE, 88(4): 451–515, April 2000.

6. D. Pan (1995), “A tutorial on

MPEG/audio compression,” IEEE

Multimedia, 2(2): 60–74.