G.711 is, by far, the most commonly supported voice companding algorithm used in telephony. It has become the de facto standard used to ensure interoperability in voice over Internet protocol (VoIP) applications. Compression is performed on a per sample basis with each uniformly quantized sample producing an 8-bit pulse code modulated (PCM) or companded value.
G.711 Appendix 1 (optional) is a highly effective algorithm for concealing lost packets of G.711 data. Voice signals are synthesized during the periods when data is unavailable for real-time playback. Using Adaptive Digital’s implementation of G.711 Appendix 1, speech remains intelligible even under conditions where up to 30% of the packets are lost.
G.711 Appendix 2 (optional) provides voice activity detection (VAD), discontinuous transmission (DTX), and comfort noise generation (CNG). Adaptive Digital’s implementation of G.711 Appendix 2 characterizes background noise in terms of both amplitude and spectral content. By transmitting this information at a very low bit rate to the receive side of the link, the synthesizer at that end is able to recreate comfort noise that mirrors both the amplitude and spectral characteristics of the original background noise. By doing so, the synthesized signal achieves a seamless, natural sound during transitions between speech and quiet portions of a conversation.
Features include:
Mu-law and a-law support
ITU G.711 Compliant
Optional voice activity detection (VAD) with discontinuous transmission (DTX) and comfort noise generation (CNG) for bandwidth reduction
Robust packet loss concealment for improved voice quality during periods of missing packets
Multi-channel capable
Functions are C-callable
Supported by ARM processor-based devices
Specifications can be viewed on our website at: http://www.adaptivedigital.com/product/arm/g.711-a1a2-arm.htm
G.711 Speech Codec with Appendices 1 & 2