G.711

主要用于电话的一种codec，G.711定义了两种算法：the µ-law algorithm (used in North America & Japan) and A-law algorithm (used in Europe and the rest of the world)。 美国和日本使用µ-law，欧洲和其他国家使用A-law。 所以VoIP电话中应该同时提供这两种codec，以实现跟全球各地的PSTN电话通话。

  • Sampling frequency 8 kHz
  • 64 kbit/s bitrate (8 kHz sampling frequency x 8 bits per sample)
  • Typical algorithmic delay is 0.125 ms, with no look-ahead delay
  • G.711 is a waveform speech coder
  • G.711 Appendix I defines a Packet Loss Concealment (PLC) algorithm to help hide transmission losses in a packetized network
  • G.711 Appendix II defines a Discontinuous Transmission (DTX) algorithm which uses Voice Activity Detection (VAD) and Comfort Noise Generation (CNG) to reduce bandwidth usage during silence periods
  • PSQM testing under ideal conditions yields Mean Opinion Scores of 4.45 for G.711 u-law, 4.45 for G.711 a-law
  • PSQM testing under network stress yields Mean Opinion Scores of 4.13 for G.711 u-law, 4.11 for G.711 a-law

 

G.722

G.722 is a ITU-T standard 7 kHz wideband speech codec operating at 48, 56 and 64 kbit/s.

 

Siren

Siren is a family of patented, transform-based, wideband audio codecs developed and licensed by PictureTel Corporation (acquired by Polycom, Inc. in 2001).[1] There are three Siren codecs: Siren 7, Siren 14 and Siren 22.

Siren 7 (or Siren7 or only Siren) provides 7 kHz audio, bit rates 16, 24, 32 kbit/s and sampling frequency 16 kHz. Siren is derived from PictureTel's PT716plus algorithm. In 1999, ITU-T approved G.722.1 recommendation, which is based on Siren 7 algorithm. It was approved after a four-year selection process involving extensive testing. G.722.1 provides only bit rates 24 and 32 kbit/s and does not support Siren 7's bit rate 16 kbit/s. The algorithm of Siren 7 is identical to its successor, G.722.1, although the data formats are slightly different.

Siren 14 (or Siren14) provides 14 kHz audio, bit rates 24, 32, 48 kbit/s for mono, 48, 64, 96 kbit/s for stereo and sampling frequency 32 kHz. Siren 14 supports stereo and mono audio. It offers 40 millisecond algorithmic delay, using 20 millisecond frame lengths. The mono version of Siren 14 became ITU-T G.722.1C (14 kHz, 24/32/48 kbps) in April 2005. The algorithm is based on transform technology, using a Modulated Lapped Transform (MLT).

Siren 22 (or Siren22) provides 22 kHz audio, sampling frequency 48 kHz, bit rates 64, 96, 128 kbit/s stereo and 32, 48, 64 kbit/s mono. Siren 22 offers 40 millisecond algorithmic delay using 20 millisecond frame lengths. In May 2008, ITU-T approved the new G.719 full-band codec which is based on Polycom Siren 22 audio technology and Ericsson’s advanced audio techniques.