The Improvement And Application Of G.723.1 In Digital AM Broadcasting System

ITU-T Recommendation G.723.1 is a dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3kbit/s. It was chosen as a source coding scheme for Digital AM Broadcasting System, because it gives good quality while compressing the audio signal to a very low bit rate. But this coder is designed to operate a digital signal obtained by first filtering the analogue input within the telephone bandwidth, then sampling at 8000 Hz and converting each sample to a 16-bit linear PCM version to the input of the encoder. The output of the decoder should be converted back to analogous audio by similar means. So the frequency range it can operate is limited to 0.3~3.4 kHz, only containing about 3 kHz in its bandwidth.However, these frequency limited audios sound "muffled" after decoding. So a method which extends the bandwidth by improving the decoder only at the receiver is proposed. Based on the speech production model, the method can be divided into two tasks: the extension of the spectra envelope of the speech signal, and the extension of its excitation signal.A method called Extension Matrix Mapping (EMM) is used for spectral envelope extension. And the excitation signal extension is performed by using a non-linear method due to its spectral flatness. Finally, the estimated wideband excitation signal is fed into the all-pole synthesis filter, in this way creating the enhanced output speech signal and then comparing their spectra.At the end of this thesis, we discuss the hardware realization of the G.723.1 CODEC based on its algorithm analysis.