改进型LPC声码器语音编码算法

本文给出了一种改进的ＬＰＣ语音编码算法，用于实现低速率声码器。与传统ＬＰＣ声码器算法相比，本算法在参数提取及合成等方面采取了一些改进措施，使得合成语音质量有很大的提高。本文在引言后概述了编码算法改进的考虑，然后给出编译码器的算法，重点讨论了本文提出的用动态规划法进行基音提取和平滑的新算法，以及合成端混合激励算法。本算法已经用ＴＭＳ３２０Ｃ２５实现单片编解码。     

1200/2400bps改进型多带激励声码器的实时实现

本文介绍了基于多带激励（ＭＢＥ）语音模型的改进型全双工１２００／２４００ｂｐｓ声码器．该声码器已应用于多种通信系统中．其语音清晰度（ＤＲＴ标准）１２００ｂｐｓ时为９１７５,２４００ｂｐｓ时为９２６７．本文重点介绍其硬件结构及算法实现．     

变码率分类型实时语音声码器

本文提出的声码器将语音分成静音、清音、浊音和混合音四类。用自适应方法进行分频带清浊音判决和有声／无声判决,提高了分类算法的稳定性、准确性和灵活性、准确性和灵活性,还保持了混合语音的音质,且无须对清浊音判决结果进行编码。对清音和浊音的频谱分别采用不同的ＬＳＰ量化表进行编码,从而用标量量化器替代子矢量量化器,降低了复杂度。声码器的码率最高２．４ｋｂｐｓ,最低为１００ｂｐｓ,平均码率１．４ｋｂｐｓ。实时     

The Residual-Excited Linear Prediction Vocoder with Transmission Rate Below 9.6 kbits/s

In this paper we present a new vocoder called the residual-excited linear prediction (RELP) vocoder. The concept of the RELP vocoder combines the advantages of linear predictive coding (LPC) and voice-excited vocoding. In the RELP system, vocal tract modeling is done by the LPC technique, and the LPC residual signal is used as the excitation signal. After low-pass filtering the residual signal is coded by adaptive delta modulation and is spectrally flattened before being fed in the LPC synthesizer. The range of the transmission rate is typically between 6 and 9.6 kbits/s; the synthetic speech in this range is quite good. As the transmission rate is lowered, the synthetic speech quality degrades very gradually. Since no pitch extraction is required, the vocoder is robust in any operating environment.     

一种2.4kbps改进型MBELP编码

该文给出了一种改进的2.4kb/s多带激励线性预测(IMBELP)语音编码算法。与传统的MBELP算法相比,本算法在基音提取和清/浊音判决上采取了一些改进措施,使得合成语音质量有一定的提高。本文详细介绍了改进后的MBELP算法,并将其在基音提取和清/浊音判决的结果与传统的MBELP进行比较。     

A Compact, Flexible LPC Vocoder Based on a Commercial Signal Processing Microcomputer

A very small, flexible, high-quality, full-duplex 2.4-kbit/s linear predictive vocoder has been implemented with commercially available integrated circuits. This fully digital realization is based on a distributed signal processing architecture employing three Nippon Electric Company (NEC) µPD7720 signal processing interface (SPI) single-chip microcomputers. One SPI implements the LPC analyzer, a second implements the Gold pitch and voicing decision algorithm, white the third µPD7720 implements the excitation generator and synthesizer. An Intel 8085-based 8-bit microcomputer is used for data transfer, control and multiplexing functions, and communications with the host terminal. The LPC chip set achieves high flexibility by accepting run time initialization options from the Intel 8085. These parameters include choice of linear predictive model (<= 15), analysis and synthesis frame size, and speech sampling frequency, as well as choice of speech input and output coding formats (linear or µ-255 law) and choice of analog or digjtal pre- and deemphasis. A total of 16 integrated circuits is used in the LPC vocoder with a power disipation of 5.5 W and occupying 18 in/sup 2/ of circuit area.     

一种基于MELP的高质量的0.6 Kb/s语音编码算法

针对极低速率语音通信的要求,提出了一种基于MELP(Mixed-Excitation Linear Prediction)的0．6Kb／s语音编码算法。把MELP算法中3个连续语音帧组成一个超级帧,充分利用参数的帧间相关性,进行联合量化,从而获得了高质量的合成语音。采用对线谱对频率的两帧联合量化与双向预测矢量量化对基音周期的按清浊音分模式量化,对子带清浊参数量化的统计码本构造,对能量参数采用分离均值矢量量化解码端对能量参数采用了一种效果更好的插值算法等。     

Implementation of a Multirate Speech Digitizer

In this paper, implementation of a compact and efficient multirate speech digitizer with variable transmission rates of 2.4, 4.8, 9.6, and 14.96 kbits/s is presented. The multirate algorithm has been made based on the residual-excited linear prediction (RELP) vocoder with a transmission rate of 9.6 kbits/s. The residual encoder employed in the RELP vocoder uses hybrid companding delta modulation (HCDM). This HCDM is also used as a 14.96 kbit/s coder. If the residual in the RELP system is down-sampled before encoding, a 4.8 kbit/s coder can be realized. If the residual encoder is not used, a 2.4 kbit/s linear predictive coder (LPC) can be realized by incorporating a pitch extractor. In the 4.8 and 9.6 kbit/s coders the pitch-implanted residual excitation method has been used to generate the excitation signal to the synthesis filter. The multirate speech digitizer algorithm has been implemented using 2900 series bit-slice microprocessors. The external memory is composed of 2K RAM's and 2K ROM's. The system design is a two-bus structure with a 204 ns cycle time. With efficient hardware and software design, the multirate speech digitizer requires almost the same hardware complexity as compared with the conventional 2.4 kblt/s LPC vocoder.     

Multiband linear predictive speech coding at very low bit rates

The authors describe the multiband linear predictive (MB-LPC) vocoder and its operation at 2.4 kb/s and 1.2 kb/s. The MB-LPC vocoder uses mixed excitation and exploits the advantages of both time and frequency domain speech coding techniques to produce natural sounding, good quality speech. Subjective performance of speech at 2.4 kb/s produced by the MB-LPC is very close to that for the 4.15 kb/s INMARSAT-M IMBE speech coder. Informal listening tests have shown that in most cases people could not tell the difference between the new 2.4 kb/s MB-LPC coder and the 4.15 kb/s INMARSAT-M IMBE coder     

Channel and source considerations of a bit-rate reduction technique for a possible wireless communications system's performance enhancement

In wireless commercial and military communications systems, where bandwidth is at a premium, robust low-bit-rate speech coders are essential. They operate at fix bit rates and those bit rates cannot be altered without major modifications in the vocoder design. A novel approach to vocoders, in order to reduce the bit rate required to transmit speech signal, is proposed. While traditional low-bit-rate vocoders code original input speech, the proposed procedure operates on the time-scale modified signal. The proposed method offers any bit rate from 2400 b/s to downwards without modifying the principle vocoder structure, which is the new NATO standard, Stanag 4591, Mixed Excitation Linear Prediction (MELP) vocoder. We consider the application of transmitting MELP-encoded speech over noisy communication channels by applying different modulation techniques, after time-scale compression is applied. Three different time-scale modification algorithms have been evaluated and waveform similarity overlap and add (WSOLA) algorithm has been selected for time-scale modification purposes. Computer simulation results, both source and channel, are presented in terms of objective speech quality metrics and informal subjective listening tests. Design parameters such as codec complexity and delay are also investigated. Simulation results lead to a possible wireless communications system, whose performance might be enhanced by using the spared bits offered by the procedure.     

低复杂度2400b／s声码器

声码器因其极低的语音压缩编码速率而得到广泛应用。但是声码器自身的内在算法特性使其通常对系统的运算量和存储量要求很高,这就为其在某些场合的应用造成了困难。为了解决以上问题,提出了一种基于正弦激励线性预测（SELP）模型的2．4kb／s低复杂度高质量声码器算法。新提出的声码器在原有SELP声码器算法基础上采取了一系列有针对性的改进措施,在大幅缩减算法时间复杂度和空间复杂度的同时,保证了很高的合成语音质量。     

基于局部余弦变换的低比特变速率语音编码算法研究

提出将局部余弦变换(LCT)算法应用于语音编码中,系统设计了一个平均比特率近1.6kbit/s的低比特变速率语音编码器。在变比特率编码器设计中采用SVM算法进行VAD检测。激活语音帧的语音模式采用GSM半速率编码中的划分方法,但将其中的强浊音模式和中浊音模式合并为一个中强浊音模式。对各类语音模式和无声帧(背景噪声)的局部余弦变换系数采用分维矢量量化算法进行量化,码书设计采用LGB算法。编码中的码书搜索采用树形快速搜索算法。通过主观非正式听力测试表明设计的变比特率编码器编码的重建语音MOS约为3.15,与比特率为2.4kbit/s美国联邦声码器标准MELP的重建语音相当,具有较强的顽健性,适合于对存在各种环境噪声的语音进行编码。     

Error correction/masking for digital voice transmission over theland mobile satellite system

The development of a 2400-b/s speech digitizer which provides an acceptable level of intelligibility and quality over land mobile satellite channels is described. Performance tests over simulated channels in the UHF band (800 MHz) are presented. The voice digitizer is a linear prediction (LPC) vocoder which uses a channel error correction and concealment procedure tailored to error statistics for a minimum-shift keyed (MSK) downlink to a moving vehicle. The error-handling technique is based on perceptual criteria and utilizes the parametric nature of LPC representation of speech. A single-error-correcting, single-burst-detecting (28, 20) fire code is shown to be the best choice for the application. The intelligibility of the vocoder is measured and compared to the standard LPC-10 algorithm. The major remaining sources of speech quality degradation due to channel errors are determined and ranked     

DPCM coding of spectral amplitudes without positive slope overload

The author considers the problem of coding the spectral amplitudes of a sinusoidal transform coding (STC) vocoder operating at 2400 b/s, which was previously considered by R.J. McAulay and T.F. Quatieri (1987). Differential pulse code modulation (DPCM) coding of spectral amplitudes in vocoders can cause formant clipping because of positive slope overload. The author presents a modified DPCM coder that prevents positive slope overload by anticipating large spectral peaks. When the vocoder is used on speech taken from a standard telephone set, a time-varying distortion of the low-end frequencies is observed. When the modified DPCM coder is applied to the telephone-set speech, its success was obvious. The time-varying distortion of the low-end frequencies is eliminated. Performance of the STC vocoder on the telephone-set speech is essentially identical to its performance on flat-frequency-response speech     

Natural quality two-band LPC coding of speech at 880 bit/s withframe interpolation

A novel frame interpolation technique for two-band linear predictive coding (LPC) vocoders is proposed for maintaining natural speech quality at bit rates below 1 kbit/s. Experimental results show that the speech quality of the proposed vocoder is quite natural at bit rates 880 bit/s and comparable to that of 4.8 kbit/s CELP     

高质量0.6 Kb/s声码器的TMS320VC55x实现

给出了一种编码速率为600b／s的高质量声码器算法及基于DSP芯片的硬件实现。介绍了语音编解码算法原理、声码器系统的硬件结构、工作流程以及软件实现与代码优化。针对C55xDSP芯片的结构特点,采用C与汇编混合编程,汇编指令优化等方法,大大降低了算法的存储复杂度和运算复杂度,达到了实时性要求。     

一种改进的2.4kb/s混合激励线性预测声码器方案

本文针对标准的2.4kb/s MELP声码器的不足之处提出了两项改进措施,一是提出了一种新的参数"能量-微分过零率比",用来对语音的过渡段和弱能量浊音段的清浊音判决进行调整;二是对线谱对的多级矢量量化(MSVQ)提出了一种多径搜索算法.实验和主观听觉测试表明,在同样2.4kb/s的码率下,改进MELP声码器的合成语音在可懂度和自然度方面都有一定的提高.     

语音增强IMBE声码器研究

多带激励声码器（ＭＢＥ）由ＭＩＴ的Ｇｒｉｆｉｎ在１９８７年提出，其改进算法（ＩＭＢＥ）已被ＩＮ－ＭＡＲＳＡＴ采纳作为卫星话音通信的标准。ＭＢＥ声码器在中低速率上可获得较好的合成语音质量，但在噪声环境中使用时，随着输入信噪比的降低，其性能将显著恶化。本文试图将语音增强技术与ＭＢＥ模型相结合以提高声码器抗噪声的性能。我们研究了两种方案：一是采用语音增强预处理器和ＩＭＢＥ声码器级联，二是将语音增强技术和ＩＭＢＥ声码器有机结合构成语音增强ＩＭＢＥ声码器。客观测试和主观试听表明，这两种系统在噪声环境中工作时，性能都有很大的提高。     

1.8kb/s的多带线性预测语音编码算法

设计了一种数码率为１．８ｋｂ／ｓ的多带线性预测（ＭＢＬＰ）语音压缩编码算法。该算法采用基于谐振结构的线性预测分析和对激励信号采用多带处理的方法。试验结果表明，本算法提供了相当于码率为２．４ｋｂ／ｓ美国联邦声码器标准ＭＥＬＰ的重建语音质量，具有较高的清晰度和自然度。     

AN IMPROVED 2.4 kb／s MIXED EXCITATION LINEAR PREDICTION VOCODER

This letter presents two improvements on 2.4 kb/s Mixed-Excitation Linear Prediction (MELP) vocoder. The one is a new parameter Redzc named energy to differential zerocrossing rate which is used in adaptation of V/UV decision of transitional segments and low energy level speech segments. The other is a multi-path searching method for Multi-Stage Vector Quantization (MSVQ) of line spectral frequency. Subjective tests show that the intelligiblity and naturallity of improved MELP vocoder are preferable to those of the original one.