| 7 kHz audio-coding within 64 kbit/s | |
 | |
| Status | In force |
|---|---|
| Year started | 1988 |
| Latest version | (09/12) September 2012 |
| Organization | ITU-T |
| Base standards | G.711 |
| Related standards | G.722.1, G.722.2, G.726 |
| Domain | audio compression |
| License | Freely available |
| Website | https://www.itu.int/rec/T-REC-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 same technology is G.726.[1]
G.722 provides improved speech quality due to a wider speech bandwidth of 50–7000 Hz compared to narrowband speech coders like G.711 which in general are optimized for POTS wireline quality of 300–3400 Hz. G.722 sample audio data at a rate of 16 kHz (using 14 bits), double that of traditional telephony interfaces, which results in superior audio quality and clarity.[2]
Other ITU-T 7 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 codecs and offers lower bit-rate compressions (24 kbit/s or 32 kbit/s). It uses a modified discrete cosine transform (MDCT) audio coding data compression algorithm.[3] A more recent G.722.2, also known as AMR-WB ("Adaptive Multirate Wideband") is based on ACELP and offers even lower bit-rate compressions (6.6 kbit/s to 23.85 kbit/s),[2] 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.[4]
- ^ "G.722 : 7 kHz audio-coding within 64 kbit/s". www.itu.int. Archived from the original on 2019-11-08. Retrieved 2019-11-15.
- ^ a b "Recommendation ITU-T G.722: 7 kHz audio-coding within 64 kbit/s". ITU-T Test Signals for Telecommunication Systems. Retrieved November 7, 2012.
- ^ Lutzky, Manfred; Schuller, Gerald; Gayer, Marc; Krämer, Ulrich; Wabnik, Stefan (May 2004). A guideline to audio codec delay (PDF). 116th AES Convention. Fraunhofer IIS. Audio Engineering Society. Retrieved 24 October 2019.
- ^ Ogunfunmi, Tokunbo; Togneri, Roberto; Narasimha, Madihally (Sim) (2014-10-14). Speech and Audio Processing for Coding, Enhancement and Recognition. Springer. p. 108. ISBN 9781493914562.