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Answer by Deepu
Submitted on 9/25/2005
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In digital signal transmission ,the analog is converted into a digital signal and after processing it may be transmitted over a limited bandwidth channel or the digital signal can be stored.If you employ signal compression the efficiency is improved. ex: In telephone communication system the BW of audio signal is around 4KHz.The signal is sampled at 8KHz and passed to A/D converter of 8 bit so the output of A/D is 64 Kbps.Using T-1 channel,we can transmit 24 signals can be transmitted using TDM.so, if the signal is compressed to 32 Kbps the number of transmitted over the same channel will be 48 Signals... The main applications is Audio Signal processing specially used in MPEG-2/4. Working Principle: The Signal x(n) is splitted into many narrow band signals which occupy contiguous frequency bands using analysis filter bank.These signals are down sampled ,yielding to sub band signals which are compressed using encoders. On the receiving side a reverse process takes place the demultiplex ,decoded up sampled and then passed through Synthesis Filter bank .. By using sub band coding efficient compression is achieved because each sub band signal can be achieved, because each sub band signal is represented by different using a different number of bits.

Answer by arun
Submitted on 2/28/2006
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Subband coder divide the signal into multiple subbands. Every subband contains a spectral part of the original spectrum. After the subband spliting every subband can be encoded separately using different resolution. Mostly uniformed subband filter are used. You split the signal into a number of equal spaced subbands. When original signal has a sample frequency of fs and you use a N time subband filter, then you have after this transformation N subbands with as sample frequency of fs/N and every subband has a spectral width of fs/(2N). Subband splitting is normally done using a MDCT with some post-processing. Computational effort is very small. Delay is typically also very small (20 ms).

Answer by prakash
Submitted on 12/1/2006
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Introduction Sub-Band Coding (SBC) is a powerful and general method of encoding audio signals efficiently. Unlike source specific methods (like LPC, which works only on speech), SBC can encode any audio signal from any source, making it ideal for music recordings, movie soundtracks, and the like. MPEG Audio is the most popular example of SBC. This document describes the basic ideas behind SBC and discusses some of the issues involved in its use. Basic Principles SBC depends on a phenomenon of the human hearing system called masking. Normal human ears are sensitive to a wide range of frequencies. However, when a lot of signal energy is present at one frequency, the ear cannot hear lower energy at nearby frequencies. We say that the louder frequency masks the softer frequencies. The louder frequency is called the masker. (Strictly speaking, what we're describing here is really called simultaneous masking (masking across frequency). There are also nonsimultaneous masking (masking across time) phenomena, as well as many other phenomena of human hearing, which we're not concerned with here. For more information about auditory perception, see the upcoming Auditory Perception OLT.) The basic idea of SBC is to save signal bandwidth by throwing away information about frequencies which are masked. The result won't be the same as the original signal, but if the computation is done right, human ears can't hear the difference. Encoding audio signals The simplest way to encode audio signals is Pulse Code Modulation (PCM), which is used on music CDs, DAT recordings, and so on. Like all digitization, PCM adds noise to the signal, which is generally undesirable. The fewer bits used in digitization, the more noise gets added. The way to keep this noise from being a problem is to use enough bits to ensure that the noise is always low enough to be masked either by the signal or by other sources of noise. This produces a high quality signal, but at a high bit rate (over 700k bps for one channel of CD audio). A lot of those bits are encoding masked portions of the signal, and are being wasted. There are more clever ways of digitizing an audio signal, which can save some of that wasted bandwidth. A classic method is nonlinear PCM, such as mu-law encoding (named after a perceptual curve in auditory perception research). This is like PCM on a logarithmic scale, and the effect is to add noise that is proportional to the signal strength. Sun's .au format for sound files is a popular example of mu-law encoding. Using 8-bit mu-law encoding would cut our one channel of CD audio down to about 350k bps, which is better but still pretty high, and is often audibly poorer quality than the original (this scheme doesn't really model masking effects).

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Question by ananth
Submitted on 10/23/2003
Related FAQ: MPEG-FAQ: multimedia compression [0/9]
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what is subband coding its use in signal processing of audio signal working principle of subband coding description of each block in subband coding applications in various fields

Answer by vajra_007
Submitted on 9/25/2004
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Answer by sbm
Submitted on 9/12/2005
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It's a coding algorithm used to decomposed signal.

Answer by pramod
Submitted on 5/3/2006
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Answer by Sri
Submitted on 11/30/2006
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SBC depends on a phenomenon of the human hearing system called masking. when a lot of signal energy is present at one frequency, the ear cannot hear lower energy at nearby frequencies. We say that the louder frequency masks the softer frequencies. The louder frequency is called the masker. The basic idea of SBC is to save signal bandwidth by throwing away information about frequencies which are masked. The result won't be the same as the original signal, but if the computation is done right, human ears can't hear the difference.

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