U. Glaeser

A new standard MPEG-7, named the multimedia content description interface, is aimed to support as
broad range of communications and multimedia applications as possible [19]. MPEG-7 audio merges
five different technologies: the audio description framework (scalable series, low-level descriptors, and
uniform silence segments), musical instrument timbre descriptors, sound recognition tools, spoken
content descriptors, and melody descriptors. In order to describe the low-level audio features, regions
of similarity and dissimilarity within the sound are searched for. This can be done either using samples
taken at regular intervals or segments of samples. The relevant samples are then further manipulated to
form a scalable series, which allows to progressively down-sample the data contained in a series, according
to the application, bandwidth, or storage requirements.
The scope of newest MPEG-21 standard is the integration of technologies enabling transparent and
augmented use of multimedia resources across a wide range of networks and devices to support functions
such as content creation, content production, content distribution, content consumption and usage, content
packaging, intellectual property management and protection, content identification and description, financial
management, user privacy, terminals and network resource abstraction, content representation and
event reporting [20].
MUSICAM as well as MPEG-1 layers I and II coders have the same structure shown in Fig. 27.31. The
input audio signal is transmitted via a 32-band polyphase analysis filter bank (Fig. 27.22(a)) with equally
spaced passbands, according to Eqs. (27.50a–c). All subband filters with impulse responses H i(n), i = 0,
1,…,31, determined by Eq. (27.48), are obtained by modulation of a single prototype lowpass filter with
the impulse response h(n), as is illustrated in Fig. 27.22(c). Their output signals are critically decimated.
For a 48 ksamples/s sampling rate, each subband filter has a passband width of 750 Hz. Although these
filters are highly overlapping, they can guarantee a perfect (or at least a nearly perfect) signal reconstruction
(via the synthesis filter bank in Fig. 27.22(b)) due to the power complementarity. For instance, at
multiples of 750 Hz the respective filters, i.e., those with neighboring passbands exhibit a 3 dB attenuation.
Samples of subband signal components are quantized with a number of uniform midtread quantizers
with 3, 5, 7,…,65,535 possible levels. Blocks of samples are formed (e.g., blocks of 12 samples in layer I)
and divided by a scalefactor s sf selected in such a way that the sample with the largest magnitude is scaled
to 1. By this means a quite large overall dynamic range of approximately 126 dB is reached. Proper
quantizers are selected with the dynamic bit allocation algorithm described in section 27.9, controlled
PCM
audio
Encoded
bit stream
FIGURE 27.31 Structure of MUSICAM and MPEG-1 layer I and II coders: (a) encoder, (b) decoder.
© 2002 by CRC Press LLC
Analysis
filter bank
(time-to-frequency
mapping)
FFT
Bit stream
unpacking
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Scaling,
bit allocation,
and
quantization
Psychoacoustic
perception
model
(a)
Dequantization
and
descaling
Ancillary data
(if included)
(b)
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Bit stream
formatting
Ancillary data
(optional)
Sythesis
filter bank
(frequency-to-time
mapping)
Encoded
bit stream
PCM audio
(reconstructed)