Editing G.722

{{short description|ITU-T recommendation}}
{{Infobox technology standard
| title             = G.722
| long_name         = 7 kHz audio-coding within 64&nbsp;kbit/s
| image             = Pcm.svg 
| caption           = 
| status            = In force
| year_started      = 1988
| version           = (09/12)
| version_date      = September 2012
| preview           = 
| preview_date      = 
| organization      = [[ITU-T]]
| committee         = 
| base_standards    = [[G.711]]
| related_standards = [[G.722.1]], [[G.722.2]], [[G.726]]
| abbreviation      = 
| domain            = [[audio compression (data)|audio compression]]
| license           =Freely available 
| website           = https://www.itu.int/rec/T-REC-G.722
}}

'''G.722''' is an [[ITU-T]] standard 7&nbsp;kHz [[wideband audio]] [[codec]] operating at 48, 56 and 64&nbsp;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 same technology is [[G.726]].<ref>{{Cite web|url=https://www.itu.int/rec/T-REC-G.722|title=G.722 : 7 kHz audio-coding within 64 kbit/s|website=www.itu.int|url-status=live|archive-url=https://web.archive.org/web/20191108201029/https://www.itu.int/rec/T-REC-G.722/|archive-date=2019-11-08|access-date=2019-11-15}}</ref>

G.722 provides improved speech quality due to a wider speech bandwidth of 50–7000&nbsp;Hz compared to narrowband speech coders like [[G.711]] which in general are optimized for [[Plain old telephone service|POTS]] wireline quality of 300–3400&nbsp;Hz. G.722 sample audio data at a rate of 16&nbsp;kHz (using 14 bits), double that of traditional telephony interfaces, which results in superior audio quality and clarity.<ref name="G series1">{{cite web|url=https://www.itu.int/net/itu-t/sigdb/speaudio/Gseries.htm#G.722|title=Recommendation ITU-T G.722: 7 kHz audio-coding within 64 kbit/s|publisher=ITU-T Test Signals for Telecommunication Systems|access-date=November 7, 2012}}</ref>

Other ITU-T 7&nbsp;kHz wideband codecs include [[G.722.1]] and [[G.722.2]]. These codecs are not variants of G.722 and they use different patented compression technologies. G.722.1 is based on [[Siren (codec)|Siren]] codecs and offers lower bit-rate compressions (24&nbsp;kbit/s or 32&nbsp;kbit/s). It uses a [[modified discrete cosine transform]] (MDCT) [[audio coding]] [[data compression]] algorithm.<ref>{{cite conference |last1=Lutzky |first1=Manfred |last2=Schuller |first2=Gerald |last3=Gayer |first3=Marc |last4=Krämer |first4=Ulrich |last5=Wabnik |first5=Stefan |title=A guideline to audio codec delay |url=https://www.iis.fraunhofer.de/content/dam/iis/de/doc/ame/conference/AES-116-Convention_guideline-to-audio-codec-delay_AES116.pdf |website=[[Fraunhofer IIS]] |conference=116th AES Convention |publisher=[[Audio Engineering Society]] |date=May 2004 |access-date=24 October 2019}}</ref> A more recent G.722.2, also known as [[Adaptive Multi-Rate Wideband|AMR-WB]] ("Adaptive Multirate Wideband") is based on [[ACELP]] and offers even lower bit-rate compressions (6.6&nbsp;kbit/s to 23.85&nbsp;kbit/s),<ref name ="G series1"/> as well as the ability to quickly adapt to varying compressions as the network topography mutates. In the latter case, bandwidth is automatically conserved when [[network congestion]] is high. When congestion returns to a normal level, a lower-compression, higher-quality bitrate is restored.<ref>{{Cite book|url=https://books.google.com/books?id=i-nSBAAAQBAJ&pg=PA108|title=Speech and Audio Processing for Coding, Enhancement and Recognition|last=Ogunfunmi|first=Tokunbo|last2=Togneri|first2=Roberto|last3=Narasimha|first3=Madihally (Sim)|date=2014-10-14|publisher=Springer|isbn=9781493914562|pages=108|language=en}}</ref>

==Applications==
G.722 is an ITU standard codec that provides 7&nbsp;kHz [[wideband audio]] at data rates from 48, 56 and 64&nbsp;kbit/s. This is useful for [[voice over IP]] applications, such as on a [[local area network]] where network bandwidth is readily available, and offers a significant improvement in speech quality over older narrowband codecs such as [[G.711]], without an excessive increase in implementation complexity. Environments where bandwidth is more constrained may prefer one of the more bitrate-efficient codecs, such as G.722.1 (Siren7) or G.722.2 (AMR-WB).

G.722 has also been widely used by radio broadcasters for sending commentary-grade audio over a single 56 or 64&nbsp;kbit/s [[ISDN]] B-channel (the [[least significant bit]] is dropped on 56&nbsp;kb circuits).

G.722 works by having the inbound voice signal pass through a digital filter that splits the audio signal into 0&nbsp;Hz-to-4&nbsp;kHz and 4&nbsp;kHz-to-8&nbsp;kHz audio bands. These sub-bands are then encoded using [[SB-ADPCM|sub-band ADPCM]]. Most of the human voice energy is concentrated in the lower half of the audio band (0–4&nbsp;kHz), so 48&nbsp;kbit/s of the bandwidth is dedicated to the lower sub-band and the other 16&nbsp;kbit/s is allocated to the higher sub-band.<ref name ="G series1"/><ref name ="cisco">{{ cite web |url=http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps8537/prod_white_paper0900aecd806fa57a.html2 |title= Wideband Audio and IP Telephony | publisher=Cisco Systems | access-date=November 7, 2012}} {{dead link|date=August 2014}}</ref>

===RTP encapsulation===
G.722 VoIP is typically carried in [[Real-time Transport Protocol|RTP]] payload type 9.<ref>{{Cite web|url=https://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml|title=Real-Time Transport Protocol (RTP) Parameters|website=www.iana.org}}</ref>
Note that [[Internet Assigned Numbers Authority|IANA]] records the clock rate for type 9 G.722 as 8&nbsp;kHz (instead of 16&nbsp;kHz), <nowiki>RFC 3551</nowiki><ref>RFC 3551 [[Request for Comments|Request For Comments]] 3551: RTP Profile for Audio and Video Conferences with Minimal Control. Schulzrinne & Casener, July 2003. Also [[Internet standard|Internet Standard]] 65.</ref> clarifies that this is due to a historical error and is retained in order to maintain backward compatibility. Consequently, correct implementations represent the value 8,000 where required but encode and decode audio at 16&nbsp;kHz.

Whilst G.722 allows for bitrates of 64, 56 and 48&nbsp;kbit/s, in practice, data is encoded at 64&nbsp;kbit/s, with bits from the lower sub-band being used to encode auxiliary data. The greater the number of bits allocated to aux data, the lower the bit rate.

e.8<ref>{{citation |url=http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-UnderstandingVoIP-SECT-3.html |title=Appendix B. Protocols for VoIP - Codecs |publisher=asteriskdocs.org }}</ref>

== See also ==
*[[List of codecs]]
*[[Comparison of audio coding formats]]
*[[Wideband audio]]

==References==
<references />

==External links==
*[https://www.itu.int/rec/T-REC-G.722/ ITU-T Recommendation G.722: 7&nbsp;kHz audio-coding within 64&nbsp;kbit/s- technical specification]

{{Compression formats}}

[[Category:Audio codecs]]
[[Category:Speech codecs]]
[[Category:Wideband codecs]]
[[Category:ITU-T recommendations]]
[[Category:ITU-T G Series Recommendations]]