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G.726 is an
ITU-T The ITU Telecommunication Standardization Sector (ITU-T) is one of the three sectors (divisions or units) of the International Telecommunication Union (ITU). It is responsible for coordinating standards for telecommunications and Information Commu ...
ADPCM 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 data bandwidth for a given signal-to-noise ratio ...
speech codec Speech coding is an application of data compression of digital audio signals containing speech. Speech coding uses speech-specific parameter estimation using audio signal processing techniques to model the speech signal, combined with generic d ...
standard covering the transmission of voice at rates of 16, 24, 32, and 40  kbit/s. It was introduced to supersede both G.721, which covered ADPCM at 32 kbit/s, and
G.723 G.723 is an ITU-T standard speech codec using extensions of G.721 providing voice quality covering 300 Hz to 3400 Hz using Adaptive Differential Pulse Code Modulation (ADPCM) to 24 and 40 kbit/s for digital circuit multiplication equip ...
, which described ADPCM for 24 and 40 kbit/s. G.726 also introduced a new 16 kbit/s rate. The four
bit rates In telecommunications and computing, bit rate (bitrate or as a variable ''R'') is the number of bits that are conveyed or processed per unit of time. The bit rate is expressed in the unit bit per second (symbol: bit/s), often in conjunction w ...
associated with G.726 are often referred to by the bit size of a
sample Sample or samples may refer to: Base meaning * Sample (statistics), a subset of a population – complete data set * Sample (signal), a digital discrete sample of a continuous analog signal * Sample (material), a specimen or small quantity of s ...
, which are 2, 3, 4, and 5-bits respectively. The corresponding wide-band codec based on the same technology is
G.722 G.722 is an ITU-T standard 7 kHz wideband audio codec operating at 48, 56 and 64 kbit/s. It was approved by ITU-T in November 1988. Technology of the codec is based on sub-band ADPCM (SB-ADPCM). The corresponding narrow-band codec based on ...
. The most commonly used mode is 32 kbit/s, which doubles the usable network capacity by using half the rate of
G.711 G.711 is a narrowband audio codec originally designed for use in telephony that provides toll-quality audio at 64 kbit/s. G.711 passes audio signals in the range of 300–3400 Hz and samples them at the rate of 8,000 samples per second ...
. It is primarily used on international trunks in the phone network and is the standard codec used in
DECT Digital enhanced cordless telecommunications (Digital European cordless telecommunications), usually known by the acronym DECT, is a standard primarily used for creating cordless telephone systems. It originated in Europe, where it is the common ...
wireless phone systems. The principal application of 24 and 16 kbit/s channels is for overload channels carrying voice in digital circuit multiplication equipment (DCME). The principal application of 40 kbit/s channels is to carry data modem signals in DCME, especially for
modems A modulator-demodulator or modem is a computer hardware device that converts data from a digital format into a format suitable for an analog transmission medium such as telephone or radio. A modem transmits data by Modulation#Digital modulati ...
operating at greater than 4800 bit/s.


History

G.721 was introduced in 1984, while
G.723 G.723 is an ITU-T standard speech codec using extensions of G.721 providing voice quality covering 300 Hz to 3400 Hz using Adaptive Differential Pulse Code Modulation (ADPCM) to 24 and 40 kbit/s for digital circuit multiplication equip ...
was introduced in 1988. They were folded into G.726 in 1990. G.727 was introduced at the same time as G.726, and includes the same bit rates, but is optimized for packet circuit multiplex equipment (PCME) environment. This is achieved by embedding 2-bit quantizer to 3-bit quantizer and same for the higher modes. This allows dropping of the
least significant bit In computing, bit numbering is the convention used to identify the bit positions in a binary number. Bit significance and indexing In computing, the least significant bit (LSB) is the bit position in a binary integer representing the binary 1 ...
from the bit stream without adverse effects on speech signal.


Features

*Sampling frequency 8 kHz *16 kbit/s, 24 kbit/s, 32 kbit/s, 40 kbit/s bit rates available *Generates a
bitstream A bitstream (or bit stream), also known as binary sequence, is a sequence of bits. A bytestream is a sequence of bytes. Typically, each byte is an 8-bit quantity, and so the term octet stream is sometimes used interchangeably. An octet may ...
, therefore frame length is determined by packetization time (typically 80 samples for 10  ms frame size) *Typical algorithmic delay is 0.125 ms, with no look-ahead delay *G.726 is a waveform speech coder which uses Adaptive Differential Pulse Code Modulation (
ADPCM 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 data bandwidth for a given signal-to-noise ratio ...
) *
PSQM Perceptual Speech Quality Measure (PSQM) is a computational and modeling algorithm defined in Recommendation ITU-T P.861 that objectively evaluates and quantifies voice quality of voice-band (300 – 3400 Hz) :Speech codecs, speech codecs. It ...
testing under ideal conditions yields
mean opinion score Mean opinion score (MOS) is a measure used in the domain of Quality of Experience and telecommunications engineering, representing overall quality of a stimulus or system. It is the arithmetic mean over all individual "values on a predefined scale t ...
s of 4.30 for G.726 (32 kbit/s), compared to 4.45 for
G.711 G.711 is a narrowband audio codec originally designed for use in telephony that provides toll-quality audio at 64 kbit/s. G.711 passes audio signals in the range of 300–3400 Hz and samples them at the rate of 8,000 samples per second ...
( μ-law) *PSQM testing under network stress yields mean opinion scores of 3.79 for G.726 (32 kbit/s), compared to 4.13 for G.711 (μ-law) * 40 kbit/s G.726 can carry 12000 bit/s and slower modem signals, while 32 kbit/s G.726 can carry 2400 bit/s and slower modem signals well and 4800 bit/s with some more degradation than clear channel codecs.


Endianness and payload type

Since the byte order for data protocols in the context of the internet was generally defined as big endian and called simply ''network byte order'', as stated (among others) by the deprecated RFC 1700, the deprecated RFC 1890 did not explicitly define the endianness of the predecessor of G.726, G.721, in RTP either. Instead of that, in the deprecated RFC 1890, the use of big endian by the term network byte order was generally stated for all mentioned codecs again: The payload type for G.721 was defined by the deprecated RFC 1890 as ''2'', thus a=rtpmap:2 G721/8000. In drafts for newer version of this RFC, it was reused for G.726, i.e. a=rtpmap:2 G726-32/8000. Contrary to that the ITU explicitly defined the byte order in its recommendations regarding G.726 or respectively ADPCM, but in two different ways. Recommendation ''X.420'' states, that it shall be little endian, respecting recommendation ''I.366.2 Annex E'' it should be big endian. This led to contradicting decisions in various implementations, as some manufacturers opted for little endian and others for big endian. The consequence was, that these implementations were incompatible, as decoding using the wrong byte order results in a heavily distorted audio signal. Therefore the unclear definition was fixed by the RFC 3551, which replaces RFC 1890. Section 4.5.4 in RFC 3551 defines the classical MIME-types G726-16, 24, 32 and 40 as little endian and introduces new MIME types for bis endian, which are AAL2-G726-16, 24, 32 and 40. The payload type was changed to dynamic, in order to prevent confusion. Instead of payload type ''2'' a dynamic payload in the range from 96 to 127 shall be used: Newer implementations respect the RFC 3551 and clearly distinct between G726-xx (little endian) and AAL2-G726-xx (big endian). The Gigaset C610 IP DECT phone, e.g., generates the following code in its SIP INVITE: a=rtpmap:96 G726-32/8000 → dynamic payload type ''96'' and G.726 according to X.420, thus little endian, as defined in RFC 3551
a=rtpmap:97 AAL2-G726-32/8000 → dynamic payload type ''97'' and G.726 according to I.366.2 Annex E, thus big endian, as defined in RFC 3551
a=rtpmap:2 G726-32/8000 → static payload type ''2'' and G.726 with unpredictable endianness, like G.721 according to the deprecated RFC 1890


See also

*
List of codecs The following is a list of compression formats and related codecs. Audio compression formats Non-compression * Linear pulse-code modulation (LPCM, generally only described as PCM) is the format for uncompressed audio in media files and it is al ...
*
Comparison of audio coding formats The following tables compare general and technical information for a variety of audio coding formats. For listening tests comparing the perceived audio quality of audio formats and codecs, see the article Codec listening test. General informatio ...


External links


ITU-T G.726 pageITU-T G.191 software tools for speech and audio coding, including G.726 C codeRFC 3551 - RTP Profile for Audio and Video Conferences with Minimal Control, G726-40, G726-32, G726-24, and G726-16
{{Compression formats Audio codecs Speech codecs ITU-T recommendations ITU-T G Series Recommendations