SAW32位抽头延迟线非相干扩频处理单元实验

本文简述了扩频通信的原理及ＳＡＷ同头延一在直接序列（ＤＳ）扩频通信技术中的应用，详细介绍了ＳＡＷ３２位抽头延迟线非相干扩频处理单元情况，该扩频处理单元使用３２位Ｍ序列ＳＡＷ抽头延迟线作为相位编码波形发生器和匹配波滤器，具有５ＭＨｓ带苋和１５６．２５ｋｂｉｔ／ｓ的数据率，在解调器中，匹配滤波器的输出经包络检波和幅度判别，以实现地扩频信号的解扩，该扩频单元具有简单和快速同步的特点。     

已颁布的推荐性标准

已颁布的推荐性标准１．２０４８ｋｂｉｔ／Ｓ数字信令转换设备技术要求和测试方法２．２０４８ｋｂｉｔ／Ｓ对称电缆再生中继设备技术条件３．３４Ｍｂｉｔ／ｓ中容量数字微波接力通信系统技术要求和测量方法４．６ＧＨＺ１４０Ｍｂ／ｓ大容量数字微波接力通信系统技术要...     

SAW32位抽头延迟线非相干扩频处理单元实验

简述了扩频通信的原理及ＳＡＷ抽头延迟线在直接序列（ＤＳ）扩频通信技术中的应用，详细介绍了ＳＡＷ３２位抽头延迟线非相干扩频处理单元实验情况。该扩频处理单元使用３２位Ｍ序列ＳＡＷ抽头延迟线作为相位编码波形发生器和匹配滤波器，具有５ＭＨｚ带宽和１５６．２５ｋｂｉｔ／ｓ的数据率。在解调器中，匹配滤波器的输出经包络检波和幅度判别，以实现对扩频信号的解扩。该扩频单元具有简单和快速同步的特点。     

SAW32位轴头延迟线非相干扩频处理单元实验

简述了扩频通信的原理及ＳＡＷ抽头延迟线在直接序列扩频通信技术中的应用，详细介绍了ＳＡＷ３２位抽头延迟线非相干扩频处理单元实验情况。该扩频处理单元使用３２位Ｍ序列ＳＡＷ抽头延迟线作为相位编码波形发生器和匹配滤波器，具有５ＭＨｚ带宽和１５６．２５ｋｂｉｔ／ｓ的数据库。在解调器中，匹配滤波器的输出经包络检波的幅度判别，以实现对扩频信号的解扩。该扩频单元具有简单和快速同步的特点。     

美国U.S.Robotics公司在普通电话线上实现56kbit/s传输

美国Ｕ．Ｓ．Ｒｏｂｏｔｉｃｓ公司在普通电话线上实现５６ｋｂｉｔ／ｓ传输世界最大的调制解调器生产厂商美国Ｕ．Ｓ．Ｒｏｂｏｔｉｃｓ公司近期在调制解调技术上实现重大突破，新推出的Ｘ２技术可以使最高模拟传输速度从目前的３４ｋｂｉｔ／ｓ提高到５６ｋｂｉｔ／ｓ，...     

通过有线电视网宽带接入Internet的研究

目前电话ｍｏｄｅｍ只能提供几十ｋｂｉｔ／ｓ的传输速率，ＩＳＤＮ接入也只有１２８ｋｂｉｔ／ｓ，使Ｉｎｔｅｒｎｅｔ用户浏览十分缓慢。作者从目前中国覆盖率最高的有线电视网出发，进行Ｉｎｔｅｒｎｅｔ宽带接入系统的研究，给出了详细的系统设计方案。     

速率为64kbit／s—1920kbit／s会议电视的帧结构

本文主要介绍速率为６４ｋｂｉｔ／ｓ－１９２０ｋｂｉｔ／ｓ会议电视在Ｎ－ＩＳＤＮ信道和ＰＣＭ一次群信道上传输的各种帧结构。     

我国发展N-ISDN的市场前景与策略

一．Ｎ－ＩＳＤＮ发展现状１．Ｎ－ＩＳＤＮ应用简介综合业务数字网（ＩＳＤＮ）是在现有电话网基础上经济有效地利用网络资源的一种技术它是采用端到端数字连接和标准的接口向用户提供包括语音、数据和图像等多媒体的综合业务。利用ＩＳＤＮ,用户可以在一条普通电话线上实现边上网边打电话、边打电话边发传真、两部电脑同时上网或者同时使用两部电话。ＩＳＤＮ有两种类型业务：ＩＳＤＮ ＢＲＩ基本速率接口：２Ｂ＋Ｄ即２Ｘ６４ｋｂｉｔ／ｓ＋１６ｋｂｉｔ／ｓ＝１４４ｋｂｉｔ／ｓ速率ＩＳＤＮ ＰＲＩ基群速率接口：欧洲标准：３０Ｂ＋Ｄ即３０Ｘ６４ｋｂｉｔ／ｓ＋６４ｋｂｉｔ／ｓ速率美国／日本标准：２３Ｂ＋Ｄ即（２３Ｘ６４ｋｂｉｔ／ｓ＋６４ｋｂｉｔ／ｓ）速率２．世界发展概况近几年由于Ｉｎｔｅｒｎｅｔ的快速发展,国外ＩＳＤＮ 业务增长的势头一直很强劲。世界上开放ＩＳＤＮ商用业务的国家１９９３年为２１个,１９９５年发展到３０个,目前遍布全球的许多国家都已能提供ＩＳＤＮ业务。１９９４年ＩＳＤＮ用户线数为１７０万,１９９５年增加到３９５万。到１９９９年,世界上基本接入（２Ｂ＋Ｄ）的用户数达１４２０万用户,美国、德国、日本用户线数居世界前３名,分别为３６...     

频率分集扩展频谱码分多址系统

频率分集扩展频谱码分多址（ＦＤ／ＳＳＭＡ，ＦｒｅｑｕｅｎｃｙＤｉｖｅｒｓｉｔｙ／ＳｐｒｅａｄＳｐｅｃｔｒｕｍＭｕｌｔｉｐｌｅＡｃｃｅｓ）系统是一种新的扩频多址通信方案。它基于多载波传输理论，并与直接序列扩频多址（ＤＳ／ＳＳＭＡ）系统在时间域和频率域构...     

一种新型速率适配数字用户线技术（RADSL）

一种新型速率适配数字用户线技术（ＲＡＤＳＬ）Ｐａｒａｄｙｎｅ公司正在推出一种“Ｇｌｏｂｅｓｐａｎ速率适配数字用户线技术（ＲＡＤＳＬ）”，宣称其速率比２８．８ｋｂｉｔ／Ｓ的调制解调器快了２００倍——下行速率为６００ｋｂｉｔ／Ｓ～７Ｍｂｉｔ／Ｓ，上行速率...     

Subjective Evaluation of Several Efficient Speech Coders

The voice quality of several 9.6 - 32 kbit/s coders is determined with an extensive set of subjective listening tests. Single encodings of μ255 PCM, adaptive differential PCM (ADPCM), subband coding (SBC), vocoder-driven adaptive transform coding (ATC), adaptive predictive coding (APC), and time domain harmonic scaling combined with SBC are compared in an idealized situation, that is, no added impairments. It is shown that single encodings of modest complexity 32 kbit/s coders such as ADPCM and SBC and more complex 24 kbit/s coders such as vocoder-driven ATC and APC offer quality nearly equivalent to 64 kbit/s μ255 PCM. However, these conclusions are drawn in the absence of a realistic telephone network where tandem encodings, delay limitations, and nonvoice signals exist. Tandem encodings of 64 kbit/s μ255 PCM, 32 kbit/s ADPCM, 16 kbit/s SBC, and 16 kbit/s APC are also evaluated. These 32 kbit/s and 16 kbit/s coders offer degraded tandem performance as compared to 64 kbit/s PCM, with the exception of synchronous tandeming of 32 kbit/s ADPCM with 64 kbit/s PCM where several encodings are subjectively equivalent to a single encoding of 32 kbit/s ADPCM.     

小波变换音频编码算法研究

任意能量有限信号都可以用紧支撑正交小波基展开或分解，这一点对研究快速高效音频编码算法是非常重要的。本文设计一种基于正交小波变换的高保真音频编码算法，该算法可以把速率为７０５．６ｋｂｉｔ／ｓ的高保真音频信号压缩到１９２ｋｂｉｔ／ｓ，１６０ｋｂｉｔ／ｓ，１２８ｋｂｉｔ／ｓ，９６ｋｂｉｔ／ｓ和６４ｋｂｉｔ／ｓ，并保持重构音频信号的高质量。     

Integration of harmonic and analysis by synthesis coders

Hybrid coding of speech has been proposed to overcome the limitations of a single model in representing the wide variety of characteristics of human speech. A new hybrid coding algorithm, which combines harmonic and analysis by synthesis coding techniques, is presented. To integrate the harmonic and analysis by synthesis coders, novel phase synchronisation and speech classification techniques are developed. The perceptual quality of the speech synthesised using the unquantised hybrid model is almost indistinguishable when compared with 128 kbit/s linear PCM. Two variable rate coders are developed based on the designed hybrid model, by quantising the parameters at different bit rates. Subjective listening tests show that the speech quality of the variable rate hybrid coders outperform the quality of 5.3 kbit/s and 6.3 kbit/s ITU G.723.1 coders, at maximum bit rates of 4 kbit/s and 6 kbit/s respectively.     

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。     

Synchronous Tandem Algorithm for 32-kbit/s ADPCM

The synchronous tandem property of nonaccumulation of distortion in tandem-connected ADPCM coders with a 64 kbit/s PCM interface is discussed here. The synchronous tandem algorithm used to provide this property in the steady-state mode is described with a case-by-case analysis, so as to show how the synchronous tandem property is realized in an ADPCM coder. A 32 kbit/s ADPCM coder utilizing this algorithm has been standardized by the CCITT (International Telegraph and Telephone Consultative Committee). The synchronous tandem property of the 32 kbit/s ADPCM coder is of great interest in network applications, because the ADPCM coder appears likely to be introduced into digital networks built partially with existing 64 kbit/s PCM circuits.     

Algorithm of Adaptive Bit Allocation Wavelet Transform Audio Coding

ＡｌｇｏｒｉｔｈｍｏｆＡｄａｐｔｉｖｅＢｉｔＡｌｏｃａｔｉｏｎＷａｖｅｌｅｔＴｒａｎｓｆｏｒｍＡｕｄｉｏＣｏｄｉｎｇＭａＨｏｎｇｆｅｉＦａｎＣｈａｎｇｘｉｎＳｏｎｇＧｕｏｘｉａｎｇ（ＸｉｄｉａｎＵｎｉｖｅｒｓｉｔｙ，Ｘｉ’ａｎ７１...     

Pitch predictive vector quantisation of speech waveform

A coding algorithm is presented which combines pitch prediction with low-dimensional vector quantisation to exploit both long- and short-term correlation in the speech waveform at rates of 16 and 9.6 kbit/s. Vector quantisation of the predictor enables the stability of the synthesis filter to be assured, and also allows the use of a minimum residual energy criterion. SNRs of 17-19 dB are achieved at 16 kbit/s and 13-15 dB at 9.6 kbit/s.<>     

An Integrated Voice/Data Terminal with Simple Synchronization Circuits Using an 80 kbit/s Ping-Pong Method

With recent digital technique progress, digitalization is spreading to subscriber loop systems. In-house systems will be digitalized earlier than other systems. In in-house networks, a pingpong method, especially an 80 kbit/s ping-pong method, using an existing cable pair, is superior to other digital transmission methods due to the sample system structure. For office use, a digital subscriber terminal is required to offer integrated services. However, the already reported 80 kbit/s method is insufficient to provide simultaneous and independent integrated services. This paper presents an 80 kbit/s ping-pong method which has 72 kbit/s capacity for the voice and data communications, so as to provide such integrated services. Furthermore, an experimental integrated terminal, which has simple synchronization circuits, is described.     

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