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From "The ISDN Studio" by Dave Immer Audio Coding ALGORITHMS For audio applications, the algorithm is a model - or a set
of rules - by which a PCM bit-stream is analyzed and re-quantized into a reduced
bit-stream. Audio coding algorithms differ in how they deal with the irrelevancy
and redundancy contained in the PCM signal and fall into two basic categories:
Transform and Predictive, both of which have subband variations. Transform is
frequency-domain based and the Predictive is time-domain based. A frequency-domain
based algorithm will employ bit reduction following known characteristics (contained
in an on-board lookup table) of human hearing. This process is called perceptual
coding and only psycho-acoustically "relevant" waveform information
is transmitted and reconstructed at the decoder where aliasing noise gets dynamically
masked within subbands of audio having the most energy at the moment. Audio
frequency response for frequency-domain coding is much less bit-rate dependent
(but has more coding delay) than a time-domain process. A time-domain approach
will use predictive analysis based on look-up tables available to the coder,
and transmit only the differences between the prediction and the actual sample
and then add the redundant information back at the decoder. The audio frequency
response is dependent on the bit-rate of the transmission but this method results
in a very low coding delay. Both approaches work quite well and each has its
advantages. A major advantage of predictive coding such as APT-X is that it
is a "near-lossless" treatment, making it a good choice for production
applications. MPEG layer 2 allows for "tweaks" and improvements in
the coding side (such as the Musicam implementation) that are "followed"
by the decoding side. This makes an MPEG layer 2 decoder much less complex (and
cheaper) than the encoder and therefore a major contender for digital radio
broadcasting. |