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Adaptive Differential Pulse

Adaptive differential pulse-code modulation (ADPCM) is a variant of differential pulse-code modulation that varies the quantization step size to further reduce bandwidth for a given signal-to-noise ratio. Typically, it uses an adaptive scale factor before quantizing the difference in the DPCM encoder. ADPCM was developed in the 1970s at Bell Labs for voice coding. In telephony, ADPCM compresses a 64 kbps PCM signal into a 32 kbps signal by mapping 8-bit PCM samples into 4-bit ADPCM samples to double line capacity. The G.726 standard details this. ADPCM is also used in Voice over IP. The G.722 standard uses subband

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142 views2 pages

Adaptive Differential Pulse

Adaptive differential pulse-code modulation (ADPCM) is a variant of differential pulse-code modulation that varies the quantization step size to further reduce bandwidth for a given signal-to-noise ratio. Typically, it uses an adaptive scale factor before quantizing the difference in the DPCM encoder. ADPCM was developed in the 1970s at Bell Labs for voice coding. In telephony, ADPCM compresses a 64 kbps PCM signal into a 32 kbps signal by mapping 8-bit PCM samples into 4-bit ADPCM samples to double line capacity. The G.726 standard details this. ADPCM is also used in Voice over IP. The G.722 standard uses subband

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srujanabhamidi
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© Attribution Non-Commercial (BY-NC)
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Adaptive differential pulse-code modulation

From Wikipedia, the free encyclopedia Adaptive differential pulse-code modulation (ADPCM) is a variant of differential pulse-code modulation (DPCM) that varies the size of the quantization step, to allow further reduction of the required bandwidth for a given signal-to-noise ratio. Typically, the adaptation to signal statistics in ADPCM consists simply of an adaptive scale factor before quantizing the difference in the DPCM encoder.[1] ADPCM was developed in the early 1970s at Bell Labs for voice coding, by P. Cummiskey, N. S. Jayant, and James L. Flanagan.[2]

[edit] In telephony
In telephony, a standard audio signal for a single phone call is encoded as 8000 analog samples per second, of 8 bits each, giving a 64 kbit/s digital signal known as DS0. The default signal compression encoding on a DS0 is either -law (mu-law) PCM (North America and Japan) or Alaw PCM (Europe and most of the rest of the world). These are logarithmic compression systems where a 13 or 14 bit linear PCM sample number is mapped into an 8 bit value. This system is described by international standard G.711. Where circuit costs are high and loss of voice quality is acceptable, it sometimes makes sense to compress the voice signal even further. An ADPCM algorithm is used to map a series of 8 bit -law (or a-law) PCM samples into a series of 4 bit ADPCM samples. In this way, the capacity of the line is doubled. The technique is detailed in the G.726 standard. Some ADPCM techniques are used in Voice over IP communications. ADPCM was also used by Interactive Multimedia Association for development of legacy audio codec known as ADPCM DVI, IMA ADPCM or DVI4, in the early 1990s.[3]

[edit] Split-band or subband ADPCM


G.722[4] is a ITU-T standard wideband speech codec operating at 48, 56 and 64 kbit/s, based on subband coding with two channels and ADPCM coding of each.[5] Before the digitization process, it catches the analog signal and divides it in frequency bands with QMF filters (quadrature mirror filters) to get two subbands of the signal. When the ADPCM bitstream of each subband is obtained, the results are multiplexed and the next step is storage or transmission of the data. The decoder has to perform the reverse process, that is, demultiplex and decode each subband of the bitstream and recombine them.

Referring to the coding process, in some applications as voice coding, the subband that includes the voice is coded with more bits than the others. It is a way to reduce the file size.

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