Refers to a suite of International Telecommunications Union (
ITU) standards for
audio (speech)
compression and decompression. This family of standards is
included in the
G Series ITU Recommendations. The G.7xx
recommendations are used in
digital transmission systems, and in particular,
used for the coding of
analog signals into digital signals.
Telephony networks employ a system of coders and decoders,
called a codec. A codec is used to
reduce bandwidth requirements over limited capacity
channels in two-way
real-time communications. The codec takes an analog voice signal from an input
source, like a microphone, and converts the analog signal into a digital
format that can be transmitted across a packet network. At the receiving end
the signal is converted back again to an analog signal for human consumption.
The G.7xx family of standards are comprised of speech and audio codecs that
are primarily used in cellular telephony and Internet telephony including
VoIP
communications. There are several specifications (protocols) in the G.7xx family
of G Series ITU Recommendations, including
the following:
G.711 - Also known as Pulse Code Modulation
(PCM), it is the ITU-T international
standard for encoding telephone audio on a 64 kbps channel. PCM
samples the
signal 8000 times a second; each sample is represented by 8
bits for a total
of 64 kbit/s. There are two versions of the this standard
codec. The µ-law
(pronounced as mew law) is generally used in North America and Japan
digital communications. The A-law is used in
European digital communications. The difference between the two standards is
the method in which the analog signal is sampled. (See also
PCM).
G.721 - An ITU-T standard codec that uses Adaptive
Differential
Pulse Code Modulation (ADPCM); a form of
Pulse Code Modulation
(PCM), to produce a digital signal with a lower bit rate than
standard PCM. This ITU standard for speech codecs uses ADPCM on a 32
kbit/s channel. NOTE: G.721 was first introduced in 1984. In 1990 this
standard was folded into G.726 along with G.723. (See also
ADPCM).
G.722 - An ITU-T standard codec that uses sub-band adaptive
differential pulse code modulation (SB-ADPCM) within a bit rate of 64
kbit/s. The system is referred to as 64
Kbps (7 kHz) audio coding.
SB-ADPCM splits the frequency band into two sub-bands (higher and lower) and
the signals in each sub-band are encoded using ADPCM. Extensions to the
G.722 standard include the following;
- G.722.1 - Is the ITU-T standard for low-complexity coding at
24 and 32 kbit/s for hands-free operation in systems with low frame
loss.
- G.722.2 - Is the ITU-T standard for coding at 24 and 32
kbit/s for hands-free operation in systems with low frame loss.
G.723 - An ITU-T standard codec that uses Adaptive Differential
Pulse Code Modulation (ADPCM) standard for
speech codecs on a 24 and 40 kbps channel. NOTE: G.723 was
first introduced in 1988. In 1990 this standard was folded into G.726 along
with G.721. (See also ADPCM).
G.726 - An ITU-T Adaptive Differential Pulse Code Modulation
(ADPCM) standard speech codec used for the
transmission of voice at rates on 16, 24, 32, and 40 kbit/s channels. G.726
supersedes both G.721 and G. 723 as it includes both of these standards plus
includes the new standard for the 16 kbit/s rate. G.726 was the standard
codec used in Digital Enhanced Cordless Telecommunications (DECT)
wireless phone systems. (See also ADPCM).
G.727 - A specialized version of the ITU-T G.726 protocol
that is intended for packet-based systems using the Packetized Voice
Protocol (PVP). G.727 uses 5, 4, 3 and 2-bit/sample embedded adaptive
Differential Pulse Code Modulation (ADPCM). (See
also ADPCM).
G.728 - An ITU-T speech coding standard that uses Low
Delay Code Excited Linear Prediction (LD-CELP) operating at 16 kbit/s
compression at a sampling rate of 8,000 samples per second. The algorithmic
coding delay of G.728 is 0.625 ms. G.728, when compared to G.726 delivers
close to the same voice quality but uses only one-half the bandwidth.
G.729 - An ITU-T audio data compression standard that operates at
8 kbit/s using a conjugate-structure algebraic-code-excited
linear-prediction (CS-ACELP). This algorithm for voice compresses voice
audio in 10 millisecond frames. G.729 is commonly used in in Voice over IP
(VoIP) applications because of its inherently low bandwidth requirement.
Extensions to the G.729 standard include the following;
- G.729a (G.729 Annex A) - Compatible with G.729 Annex A
specifies a coder with several simplifications, including code book
search routines. These modifications are known to often result in a
slightly lower voice quality.
- G.729b (G.729 Annex B) - Compatible with G.729, Annex B
specifies a coder that uses Discontinuous Transmission (DTX), Voice
Activity Detection (VAD), and Comfort Noise Generation (CNG) to reduce
bandwidth usage. Bandwidth is reduced by preventing the transmission of
non-voice during periods of silence.
