Speech coding in mobile radio communications

Speech coding, the efficient representation of speech in digital form, is one of the key technologies in current and evolving digital cellular and wireless voice communications offerings. The speech coders in existing standards exhibit a level of sophistication and performance unimaginable just 15 years ago. We outline the characteristics of the mobile communications problem with respect to speech coders and point out the principal issues in speech coder design for these applications. Speech coding methods in existing mobile communications standards are described and contrasted. The limitations imposed by the wireless channel and by background impairments are discussed, and approaches to addressing their resulting effects are presented. Suggestions for future research in speech coding for the mobile communications problem are outlined     

High-quality coding of telephone speech and wideband audio

Digital speech technology is reviewed, with the emphasis on applications demanding high-quality reproduction of the speech signal. Examples of such applications are network telephony, ISDN terminals for audio teleconferencing, and systems for the storage of audio signals, which include the important subclass of wideband speech. Depending on the application, the bandwidth of input speech can vary from about 3 kHz to nearly 20 kHz. Coding for digital telephony at 4 and 8 kb/s, network quality coding at 16 kb/s, and coding for audio at 7 and 20 kHz are examined. Future directions in the field are discussed with respect to anticipated technology applications and the algorithms needed to support these technologies     

Compact helium speech unscrambler using charge-transfer devices

Helium speech is the term commonly used for the distorted speech uttered by deep-sea divers breathing in a helium/oxygen mixture. Present unscrambler designs use pitch synchronous time-expansion signal processing with digital storage. The compact unscrambler reported here has been configured using analogue charge-transfer devices for waveform storage and c.m.o.s. digital circuitry for control logic as a precursor to development of the whole system as a single integrated circuit. The compact unscrambler itself is shown to offer distinct engineering and operational advantages.     

Ensuring mobile communication transmission quality in the year 2000

The traditional view of mobile telecommunications is to allow a customer to communicate away from his base. However, the vision for the end of the century is to make the mobile communication system the first choice for telecommunications. By the year 2000 there will be a large number of mobile systems operating in Europe, from the current analogue cellular systems to digital GSM cellular systems and potentially the Universal Mobile Telecommunication System (UMTS). The challenge is to ensure that customers, be they fixed or mobile, are provided with acceptable speech quality, no matter what system they are using and what it is interconnecting to. The paper examines the speech transmission parameters which effect the perceived quality and shows how they are apportioned within the mobile systems. It then goes on to show how the transmission quality can be optimised through the transmission plan to ensure that the customer perceives acceptable speech quality     

Digital representations of speech signals

This paper presents several digital signal processing methods for representing speech. Included among the representations are simple waveform coding methods; time domain techniques; frequency domain representations; nonlinear or homomorphic methods; and finaIly linear predictive coding techniques. The advantages and disadvantages of each of these representations for various speech processing applications are discussed.     

数字跳频系统CVSD编码性能优化研究

为了解决在受阻塞干扰的跳频信道上语音质量受影响严重的问题,对以传输语音为主的数字跳频系统采用CVSD(连续可变斜率增量调制)编码时的编码算法进行了优化研究。在传统CVSD编码基本原理的基础上,提出了一种新的CVSD编码优化算法,并给出了数字跳频系统中语音质量衡量准则和基于优化算法的数字跳频系统CVSD基带仿真模型,具体分析了在部分频带噪声干扰下优化算法对系统接收语音信号恶化量的影响。仿真结果表明,优化算法较之传统CVSD编码算法能有效地控制接收语音信号恶化量,即使在受阻塞干扰十分严重的情况下也能获得较好的语音质量。     

Techniques for improving the performance of CELP-type speech coders

Techniques for improving the performance of CELP (code excited linear prediction)-type speech coders while maintaining reasonable computational complexity are explored. A harmonic noise weighting function, which enhances the perceptual quality of the processed speech, is introduced. The combination of harmonic noise weighting and subsample pitch lag resolution significantly improves the coder performance for voiced speech. Strategies for reducing the speech coder's data rate, while maintaining speech quality, are presented. These include a method for efficient encoding of the long-term predictor lags, utilization of multiple gain vector quantizers, and a multimode definition of the speech coder frame. A 5.9-kb/s VSELP speech coder that incorporates these features is described. Complexity reduction techniques which allow the coder to be implemented using a single fixed-point DSP (digital signal processor) are discussed     

Entropy coding in speech and television differential PCM systems (Corresp.)

Much of the redundancy in a speech or television signal is eliminated when it is encoded into digital form by a differential pulse-code-modulation (DPCM) encoder. Additional coding of the DPCM output using entropy coding techniques (Huffman or Shannon-Fano coding) can result in a further increase in the signal-to-quantizing-noise ratio of 5.6 dB without increasing the transmission rate.     

Wideband speech and audio coding

Typical parameters of wideband speech and audio signals, including digitized versions of each, potential applications, and available transmission media, are described. Facts about human auditory perception that are exploited in audio coding and quality measures that play an important role in coder evaluations and designs are reviewed. Techniques for efficient coding of wideband speech and audio signals, with an emphasis on existing standards, are discussed. The audio coding standard developed by the Moving Pictures Expert Group within the International Organization for standardization (ISO/MPEG) is covered in some detail, since it will be used in many application areas, including digital storage, transmission, and broadcasting of audio-only signals and audiovisual applications such as videotelephony, videoconferencing, and TV broadcasting. Ongoing research and standardization work is outlined     

A nonlinear adaptive estimation method based on local approximation

One of the most important problems in signal processing is to estimate the output for a query from the input/output (I/O) data seen so far. This paper presents a nonlinear adaptive estimation method based on the n-nearest neighbor approach. In this method, observed I/O data are stored in a database in the form of a X-dimensional binary digital search trie (k-D trie), and a nonlinear local model to answer each query is derived based on regularization theory. The database contents are efficiently time updated to follow nonstationary data. A storage procedure allowing a simple and efficient update is developed for reduction in processing time and storage requirement. The effectiveness of the proposed method is demonstrated with both simulation data and real speech signals     

PWM方式输出合成语音

针对采用波形编码方式语音合成集成电路设计出的新型D/A转换方式,即利用脉冲宽度调制（PWM）技术,将数字语音信息直接转化成脉冲宽度调制波,通过低通滤波器可以恢复出模拟语音信号。本文所设计的数字脉冲宽度调制器采用CMOS工艺,适合于不同的语音合成集成电路。     

Speech recognition on a distributed array processor

A highly parallel single-instruction multiple-data (SIMD) array signal processor is advocated as efficient for a wide range of real-time problems. We examine its performance for digital speech recognition and show that impressive throughput rates for realistic vocabulary sizes can be achieved for `time-warping? dynamic programming algorithms which currently form the basis of several commercial and research speech recognisers.     

Digital audio coding for visual communications

Current and future visual communications for applications such as broadcasting videotelephony, video- and audiographic-conferencing, and interactive multimedia services assume a substantial audio component. Even text, graphics, fax, still images, email documents, etc. will gain from voice annotation and audio clips. A wide range of speech, wideband speech, and wideband audio coders is available for such applications. In the context of audiovisual communications, the quality of telephone-bandwidth speech is acceptable for some videotelephony and videoconferencing services. Higher bandwidths (wideband speech) may be necessary to improve the intelligibility and naturalness of speech. High quality audio coding including multichannel audio will be necessary in advanced digital TV and multimedia services. This paper explains basic approaches to speech, wideband speech, and audio bit rate compressions in audiovisual communications. These signal classes differ in bandwidth, dynamic range, and in listener expectation of offered quality. It will become obvious that the use of our knowledge of auditory perception helps minimizing perception of coding artifacts and leads to efficient low bit rate coding algorithms which can achieve substantially more compression than was thought possible only a few years ago. The paper concentrates on worldwide source coding standards beneficial for consumers, service providers, and manufacturers     

VLSI architecture for digital processing of speech signals

Due to the evolution of increasingly high performantdsp algorithms for bit rate reduction of speech signals in telecommunications,vlsi implementations of these applications are becoming more and more complex. The solutions currently being used for these applications are general purpose digital signal processors or dsp cores which are never fully adapted to the application in terms ofvlsi architecture, i.e. silicon area and power consumption. We propose an alternative to these solutions, based on a parametrable Harvard architecture, and a C compiler which gives an optimized microcode suited to this architecture and to the application. Finally we present two examples of audio applications implemented using this solution.     

Vocoders: Analysis and synthesis of speech

Techniques for analysis and synthesis of speech signals are reviewed with emphasis on vocoders and related devices for more efficient transmission and storage of speech. Selected applications of speech coding methods as sensory aids to the handicapped are described.     

基于MCP的快速光子计数成像系统的设计方案

通常基于MCP的阳极探测系统获得图像的速度受到噪声和信号处理速度的限制,影响了图像的时间分辨率和空间分辨率。介绍了一种快速光子计数成像探测器,使用的阳极结构避免了需要高信噪比的信号测量电路,及模拟信号到数字信号的转换,缩短了电子信号处理的时间,能有效提高图像质量。阳极面板依据格雷码编码方式划分区域,因此从微通道板出来的电子云在位敏阳极上的质心位置,由成对电极上的电荷比较值来确定,省略了由模拟信号变为数字信号的过程。所以对比较器的要求较高,要有高的运算速度,前端电子学具有高速、低噪、线性的特性。在此对快速光子计数成像系统提出了整体设计方案,并得到初步的实验结果,证实方案可行。     

数字音频压缩编码技术及其应用

音频信号数字化后的数据量相当大，不利于传输和存储，在日益广泛应用的数字技术中，数字音频的压缩成为其中的关键技术之一。本文先简要介绍了几种非压缩数字音频格式，随后，阐述音频压缩编码技术的发展现状，并分析了国内外现存的一些商用音频压缩系统，如G系列数字音频压缩标准、MPEG系列伴音标准及杜比数码系列和各自不同的应用领域。最后，对数字音频压缩编码技术进行了展望。     

语音合成ASIC的原理与设计方法

本文阐述了语音合成专用集成电路(ASIC)的数字信号处理原理.采用波形自适应增量调制编码方法,研究并解决了语音信号的压缩与加密.用半定制标准单元CAD设计方法,CMOS工艺,完成了32K ROM 6秒语音ASIC芯片的设计.     

数字集群通信系统中的变速率语音编码

变速率语音编码技术对于提高移动通信系统的话音服务质量起着十分关键的作用。论文首先介绍变速率语音编码技术的基本原理和相关技术,然后阐述数字集群通信系统中所采用的变速率语音编码方法,最后展望了数字集群领域变速率语音编码的发展方向。     

Moving pictures-image processing for telecommunications

The idea of providing moving colour picture videophone services, better audio quality and new data services has long been one of the dearest dreams of all telecommunications network operators. Recent advances in image and speech processing, and in implementation and network technologies, are at last coming together to make this dream a reality in the very near future. Here, the author examines the principles of the transmission of television signals in a digital form, and describes how, with radio-spectrum space at a premium, the use of telecommunications lines to transmit pictures is becoming ever more attractive