27-01-2012, 02:03 PM
Abstract
This paper presents a sufficiently low bit rate Hi-Fi
audio coding technique with low computation designed
for transmitting real-time high-quality audio signal over
wireless channel. This technique applies wavelet packet
transform to decompose audio signal into subbands to
eliminate redundant data using spectral and temporal
masking properties. The encoded audio data is framed
with some critical field is protected by channel coding to
improve noise immunity when frames are transmitted
wirelessly. Experimental results show that transparent
CD-audio quality can be achieved at 80kbps encoding
bit rate. Moreover, the proposed technique still offers
near CD-audio quality when frames are transmitted over
AWGN channel with BER below 10-5. These encouraging
results clearly exhibit the superior features of our
technique compared to others such as Ogg/Vorbis and
MP3, which are ubiquitously employed nowadays.
Keywords: hi-fi audio coding, wavelet packet transform.
1. Introduction
Although wireless communication has played a great
role in our lifestyle but to transmit high fidelity (a.k.a.
hi-fi) audio signal wirelessly at a reasonable cost is still
challenging. Presently available audio coding techniques
aim to reduce bit rate and put less concern on complexity
and efficient wireless transmission. Such audio CODECs
like ISO/MPEG [1,2,3] and Ogg/Vorbis [4] are suitable
for non real−time applications and audio archive. On the
other hand, this research focuses on a hi-fi (near CD
quality) audio coding technique that provides sufficiently
low bit rate and complexity (hence, can be processed in
real−time). In addition, the technique must tolerate the
noise (i.e., bit error rate—BER) in wireless channel at a
reasonable degree.
Current hi-fi audio CODECs employ entropy coding
such as run-length and Huffman code [3,5,6], where
important parameters required to decode are assumed to
be error−free. Otherwise, the frame will be discarded
making it susceptible to noisy wireless channel. As a
result, we identify such parameters and protect them
using channel coding that can correct up to 29 bits (one
of every seven bits). Perceptual coding technique is used
reduce the bit rate based on “human hearing masking”
property. In general, perceptual codec consists of five
modules which are filer bank, psychoacoustic analyzer,
bit allocation, quantizing and encoding, and framing, as
shown in Figure 1.
Figure 1 Structure of Perceptual Codec
The output from filter bank (time/frequency analysis)
is quantized according to masking thresholds calculated
by the psychoacoustic analyzer. In [1,2] use polyphase
filter bank which requires 512 coefficients to represent
each filter. This can take significant time to encode the
signal. In contrast, we use wavelet−based filter bank to
transform signal into wavelet domain that analyzes both
time and frequency simultaneously. The filter bank uses
fewer coefficients and can represent variable sized subbands
that more accurately match the characteristic of
non−stationary audio signal [7]; human can only detect
frequency difference at low− or medium−frequency.
This hypothesis leads us to believe that the proposed
technique should be simpler (can be implemented in
hardware). Moreover, MATLAB experiments confirm
that, at 80kbps, our wavelet−packet audio codec yields
comparable audio quality to that of the 64kbps MP3 and
Ogg/Vorbis. It is worth noting that the higher bit rate of
wavelet codec partially accounts for channel coding.
This makes it wireless transmission ready.
The rest of the paper is organized as follow: Section 2
elaborates on structure, function, and algorithm of the
proposed wavelet−packet audio codec. Experimental
results are shown in Section 4. Section 5 concludes and
presents future work.
