Low-delay vector excitation coding of speech at 16 kb/s

The need for a 16-kb/s speech coding algorithm that has very low coding delay while achieving essentially the same high quality as the 32-kb/s adaptive differential pulse code modulation (ADPCM) standard G.721 is addressed. The authors describe low-delay vector excitation coding (LD-VXC), a new coding algorithm which provides high quality with less than 2 ms of coding delay and is robust to transmission errors. The algorithm combines techniques such as vector quantization, analysis-by-synthesis, and perceptual weighting together with backward adaptive linear predictive encoding, and uses a novel long-term predictor employing backward adaptive pitch tracking. Perceptually based nose shaping and postfiltering contribute to the masking of audible quantization noise     

A low delay 16 kb/s speech coder

A code tree generated by a stochastically populated innovations tree with a backward adaptive gain and backward adaptive synthesis filters is considered. The synthesis configuration uses a cascade of two all-pole filters: a pitch (long time delay) filter followed by a formant (short time delay) filter. Both filters are updated using backward adaptation. The formant predictor is updated using an adaptive lattice algorithm. The multipath (M, L) search algorithm is used to encode the speech. A frequency-weighted error measure is used to reduce the perceptual loudness of the quantization noise. The addition of the pitch filter gives 2-10-dB increase in segSNR (segmental signal-to-noise ratio) in the voiced segments. Subjective testing has shown that the coder attains a subjective quality equivalent to 7 b/sample log-PCM (pulse code modulation) with an encoding delay of eight samples (1 ms with an 8-kHz sampling rate)     

Low-delay hybrid vector excitation linear predictive speech coding

A hybrid approach in determining the excitation vector in a low-delay code excited linear predictive coder is proposed. By a judicious division of the composite excitation vector into long-term and short-term components, and the use of switched quantisation, substantial improvement in coding quality is obtained.<>     

Application of 32 and 16 kb/s speech encoding techniques to digital satellite communications

This paper describes the performance of various voice encoding techniques at 32 and 16 kb/s for applying to digital satellite communication systems. The subjective performances of adaptive differential PCM (ADPCM), adaptive predictive coding (APC), subband coding (SBC) and adaptive delta modulation (ADM) are compared under various satellite channel environments, that is, random and burst channel errors in satellite link and an ambient noise in the ship-to-shore direction in a maritime satellite channel. The performance of the voiceband data at 4·8 and 2·4 kb/s is also evaluated for these coders. ADPCM encoding at 32 kb/s is very attractive for conventional fixed satellite systems, keeping the equivalent quality to 64 kb/s PCM. On the other hand, APC encoding at 16 kb/s is also most suitable for maritime satellite communication systems at the sacrifice of a small degradation of speech quality.     

Vector-quantised transform coder for speech coding at 9-6 kbit/s and below

The transform approach to speech coding has been established for some time, and has been shown to be very efficient in controlling the bit allocation and the shape of the noise spectrum. Various transform coders have been reported which produce high-quality digital speech at around 16 kbit/s. Although these coders can maintain good quality down to about 9.6 kbit/s, they perform poorly at lower bit rates. Here we discuss how vector quantisation (VQ) can be used to improve the quality of transform coders. We describe one specific design of vector-quantised transform coder (VQTC) which follows on from earlier work, and which is capable of producing good-quality speech at as low as 4.8 kbit/s.     

Performance of ITU-T G.729 8 kb/s CS-ACELP speech codec withnonvoice narrowband signals

A report is given on the results of a series of objective measurements conducted by COMSAT in a laboratory environment aimed at characterizing the narrowband performance of the ITU-T G.729 8 kb/s conjugate-structure algebraic code-excited linear prediction (CS-ACELP) speech coder. The test procedures followed ITU-T Recommendation G.720, “Characterization of Low-Rate Voice Coder Performance with Non-Voice Signals”. It was concluded that the G.729 algorithm has excellent performance with narrowband signals in general (e.g., single tones and DTMF signals). As for Signaling System No. 5 (SS5) interregister signals, the G.729 CS-ACELP frequently failed to correctly identify SS5 digit 6 in a number of occurrences, using worst-case analysis equipment. This indicates that the SS5 performance of G.729 codecs in trunks where SS5 is used should be carefully measured before the network planner decides on its deployment. Great care should also be taken for tandem connections, since no test has been performed for these configurations     

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.     

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     

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。     

基于小波变换的2.4kbit/s波形内插语音编码算法

基于双正交小波滤波器组对波形内插编码中提取的特征波进行多级分解与重构，提出了一种基于小波变换（WT）的2．4kbit／s特征波形内插（CWI）语音编码算法。编码端去除了特征波对齐运算，并对幅度谱进行多级分解，相位谱不传输，鉴于小波变换对信号的压缩特性，仅传输对人耳感知起主要贡献的最后一级特征波幅度谱；解码端对各尺度空间采用单独重建的方法，相位信息在重构的末级与幅度谱结合，并由浊音度标志选择固定或随机相位。此外，根据语音信号的时变特性，由基于子帧的浊音度标志选择需要传输的幅度谱及量化模式。主观R-A／B测试表明，这种基于小波变换的2．4kbit／s编码算法的合成语音质量明显优于标准的2．4kbit／s的MELP编码器及FS1016的4．8kbit／sCELP编码器，亦优于3．8kbit／s的传统CWI编码框架下的合成语音效果。     

一种基于予帧联合编码的600b/s低速语音编码算法

为了适应无线通信等甚低速语音通信应用,文中提出一种基于子帧联合编码的600b/s语音编码算法。该算法的激励源采用混合激励模型,声道参数使用帧内预测多级矢量量化进行高效量化,在参数编解码时提出了子帧分类联合的思想,并在编码端使用语音增强仰制背景噪声,解码端使用后滤波处理来改善语音质量,这些方面较传统LPC算法有了明显改进。同时,选用TI公司的TMS320VC5416DSP芯片实时实现了该算法。非正式主观试听结果表明,该算法在可懂度、消晰度等方面与传统的2.4kb/sLPC自法相当,而速率仅为LPC算法的1/4,是甚低速率的一种良好的编码方案。     

APC-AB codec modules operating at 16 and 8 kbit/s

High-quality speech codec modules operating at 16 and 8 kb/s have been developed using an adaptive predictive coding with adaptive bit allocation (APC-AB) scheme. An optimized APC-AB algorithm is studied that reduces processing complexity while maintaining speech quality. The coding algorithm is implemented in two digital signal processors (DSPs). The DSP chips, a framing LSI circuit, a PCM codec, and some peripheral ICs are integrated in each of two compact packages, i.e. codec modules, operating at 16 or 8 kb/s. The codec module size is as small as 80 mm×50 mm×12 mm, and its typical power consumption is 500 mW using 2-μm CMOS LSI technology. At 16 kb/s this APC-AB codec achieves high speech quality, close to that of a 7-bit μ-law PCM. The codec modules are expected to be used for various applications such as customer premises multiplexers for digital leased lines, digital mobile radio, and stored-and-forward-message systems (voice-mail systems)     

The road to G.729: ITU 8-kb/s speech coding algorithm with wirelinequality

In November 1995 the International Telecommunication Union Telecommunications Sector (ITU-T) approved an 8-kb/s speech coding algorithm with wireline quality. This culminated the effort that the CCITT had set in motion in 1990. This article presents the methods for managing the project through its major milestones from setting the terms of reference to the selection, testing, optimization, and dissemination of the algorithm. While G.729 was being finalized, a new requirement for a low complexity 8-b/s speech coding arose. This article explains how the change in scope was accommodated without the unnecessary proliferation of incompatible algorithms     

A REAL-TIME IMPLEMENTATION OF 4.2Kb/s CELP SPEECH CODING

This paper presents a real-time implementation of 4.2Kb/s CELP speech coding on single DSP chip. An algorithm reducing search complexity for adaptive codebook is suggested; the solving method that the parameters are changed into LSP parameters is discussed. The realtime implementation process of this coding on a commercial development board with a single TMS320C30 is described.     

一种改进的4.8kb／s码激励线性预测语音编码

本文介绍了码激励线性预测（ＣＥＬＰ）语音编码的基本原理，研究了一种制约随机激励的线性预测编码方案。它将随机激励码字进入合成滤波器的数量与自适应码本的性能指标联系起来，有效地减少了激励噪声对合成语音的影响。计算机模拟结果表明，这种方法在主观上改善了语音质量。     

Investigation of 9.6 kb/s data transmission via a pcm link at 64 kb/s with and without link errors

The problem of pulse code modulation (PCM) bit errors causing voice frequency (VF) modem errors has been studied in detail. The error mechanism consists of the addition of an impulse response type error signal added to the reconstructed VF data signal waveform at the digital-to-analogue (D/A) output of the PCM decoder. This error signal may cause a burst of errors registering in the VF modem receiver, depending on which PCM bit is in error. Numerous data have been collected in a laboratory experiment and analysed in various ways. The average bit error rate (BER) enhancement factor of VF data over PCM is between 10 and 20. For each PCM bit in error, an average of two VF data bits are in error. The analysis leads to suggestions for possible solutions to the problem.     

Transform coding of speech using a weighted vector quantizer

A medium-band speech coder is proposed that uses a weighted vector quantization scheme in the transformed domain. The linear prediction residue is transformed and vector-quantized. In order to control the quantization errors in the transformed domain, adaptively weighted matching is used instead of conventional adaptive bit allocation. Therefore, the residual signal can be reconstructed by the decoder, even if the spectral envelope parameters are destroyed due to transmission errors. This coder is also capable of maintaining higher SNR (signal-to-noise ratio) performance than time-domain vector quantization coders for a wide range of computation complexities and bit rates. Coded speech is natural and unaffected by background noise. The mean opinion score for this coder at 7.2 kb/s is comparable to that of 5.5-bit log PCM coded speech sampled at 6.4 kHz