| IntroductionIn mobile, satellite and military communications systems, the technology of speech coding plays an important role in increasing the availability of the channel by compressing transmission bandwidth and reducing transmission bit-rate of the speech signal. In recent years, the technology of speech coding advances rapidly. With the development of the signal processing and communication technology, the focus of speech coding research is centralized on the study and realization of low and very low bite rate speech coding algorithms.The traditional LPC vocoder is too simple for the speech signal model which partitions unvoice and voice in whole spectrum of the speech, so that the synthetical speech lacks the naturalness and robustness. The code-excited LPC algorithm(CELP)constructs an LPC excitation signal by the least rule of perceptual weighted error choose vectors from two codebooks: an "adaptive" codebook and a "stochastic" codebook, the algorithm can get highly synthesized speech quality, but the coding bit-rate continue to drop will result in the fast descend of the synthesized speech quality. In recent years, the representative speech coding algorithms have AMBE, MELP, WI, STC and so on in the study of equal to or less than 4kbps speech coding. These algorithms not only largely reduce the coding rate, but also economize bandwidth. The MELP is an good algorithm in current low bit rate speech coding. MELP coder has been adopted as the new US Federal Standard at 2.4kbps. the algorithm combine the merit of LPC and MBE algorithm. Several careful research and many experiments on the aspects of speech analysis, parameter code/encode and speech synthesis have been carried out. some new methods and ameliorations are employed in pitch detection, vector quantization and transmittion of LPC parameters. An improved 1.8 kbps MELP coding algorithm is proposed . 一,An Improved MELP Low Bit Rate Speech Coding Algorithm1. MELP Model The MELP coder is based on the LPC model with additional features including mixed excitation, aperiodic pulses, adaptive spectral enhancement, pulse dispersion filtering, and Fourier magnitude modeling. These additional parameters largely amend the excitation structure of the LPC model, at the same time eliminate mechanical tone noise that come forth in LPC speech synthesize. these allow the MELP vocoder to simulate accurately natural speech.At this way MELP vocoder can synthesize the high quality speech. It has become one of the best potential low bit rate speech coding.Differing from LPC10 simple unvoice/voice distinguish, MELP vocoder adopt mixed excitation. Each frame is divided into five bands and U/V determination is made in every band. The five subband signals of the speech were summed up yielding the mixed excitation, it reduce the humming of the LPC vocoder. When the input signal is voiced, MELP encoder can synthesize the speech by the cycle or the aperiodic pulse. The aperiodic pulse is mostly used in U/V conversion speech. It can form the irregular glottis pulse without introducing other tones.The adaptive spectral enhancement filter is a zero/pole filter, it make the synthetical speech and the natural speech match on better wave forms in the resonance district.pulse dispersion filtering deal with the synthetical speech by a regular pulse shaping filter. It can make the excited signal energy scatter on the whole pitch. It make the synthetical speech and the natural speech match on better wave forms in the unresonance district, contribute to dispelling some ear-piercing noise.In code part, fourier transform is used in the residual signal through LPC inversely filtering, and adopt the first 10 harmonic factor, and passed to the decode after the quantization, used to synthesize the cycle pulse, contribute to improving the naturalness of the synthetical speech, especially in male voice and backgroud noise. 2. Speech AnalysisThe input signal passes the pretreatment at first, through the high-pass filter of 60Hz, with the purpose of suppressing 50Hz p...
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