g.723.1

The g.723.1 is a dual-rate voice encoder, which is a compression algorithm recommended by ITU-T for voice or other audio signals in low-rate multimedia services, and its target application system includes h.324 and other multimedia communication systems, the algorithm has become one of the required algorithms in IP telephony system, the length of the encoder frame is 30ms, and the 7.5ms is forward-looking, the encoder's algorithm delay is 37.5ms; The encoder first makes the traditional telephone bandwidth filtering (based on g.712) for the speech signal, and then the traditional 8000- The Hz rate is sampled (based on g.711) and transformed into a linear PCM code as input to the encoder, the output is reversed in the decoder to reconstruct the speech signal, and the high-speed encoder uses the multi-pulse maximum likelihood quantization (MP-MLQ), The low rate encoder uses the generation of the digital excitation linear prediction (ACELP) method, both the encoder and the decoder must support both rates, and can convert the two rates between frames, and the system can also compress and decompress music and other audio signals, but it is optimal for voice signals The use of silent compression to perform discontinuous transmissions means that artificial noise is added to the bit stream during silence. In addition to the reserved bandwidth, this technique keeps the modem of the transmitter in continuous operation and avoids the Shitong of the carrier signal.

? Sample rate: 8kHz

? Frame Length: 30ms

Copyright NOTICE: This article for Bo Master original article, without Bo Master permission not reproduced.

g.723.1