The Design of an ADPCM/TASI System for PCM Speech Compression

This paper describes the design of a digital speech interpolation (DSI) system called ADPCM/TASI for adaptive differential PCM with time assignment speech interpolation. This system is designed to compress the output of two T1 24-channel PCM carrier terminals into a 1.544 Mbit/s signal that can be transmitted over a single T1 carrier line. The design is based on a bit slice microprocessor structure. Alternative designs are also described.     

The Tradeoff Between Delay and TASI Advantage in a Packetized Speech Multiplexer

A packetized speech multiplexer differs from a circuitswitched TASI system in that the presence of a packet buffer allows a tradeoff where the TASI advantage can be increased at a cost in packet delay. This tradeoff is investigated via a simulation. Results are presented to show the relations between TASI advantage and delay, for both an average delay criterion and a maximum delay criterion. It is shown that, particularly for the case where small numbers of talkers are multiplexed, the packetized system offers significant improvements in TASI advantage over the conventional circuit-switched multiplexer, at modest costs in packet delay.     

Fractional Speech Loss and Talker Activity Model for TASI and for Packet-Switched Speech

The cutout fraction in a TASI system is shown to bephi=frac{1}{np}sum_{k=c+1}^n (k-c) nchoose k p^{k}(1-p)^{n-k}wherenis the number of sources,cis the number of channels, andpis the probability that a source is issuing a talkspurt at a random time. This result is shown to hold independently of the probability density function of talkspurt duration. The same formula is shown to apply to the fraction of packets lost in a packet-switched link with a transmission capacity ofcpackets every Tpseconds, where Tpis the interval between packet generations for an individual source during talkspurt, and where no packet is queued for a time longer than Tp. In addition, a simple Markov birth-death model is presented for the random processa(t)representing the number of talkers issuing talkspurts at a given time.     

语音合成系统的超大容量存贮问题

文中提出了语音合成技术实现时的超大容量存贮问题 ,简要介绍了语音参数合成法中较优秀的线性预测编码方法 ,指出了它在实际应用中存在的困难 ,介绍了美国TI公司两种改进的预测编码算法MELP和CELP及相应硬件MSP50x3x系列语音合成器的原理 ,最后提出了一个应用系统实例 ,给出了硬件和软件设计。     

USB实时语音存储方法在DSP系统中的实现

以DSP芯片TMS320VC5509A为主控制器、CH376为USB接口芯片构建了实时语音存储系统。实现了DSP的USB主机功能,可以将语音数据通过USB接口以文件的形式存储到便携的USB移动存储设备中。在设计系统软件时为了保证在采集、处理与存储的过程中不丢失语音样点,采用了准多线程的处理技术和多缓冲区数据存储模式。实验证明:在不影响实时语音采集的同时,往USB移动存储设备中存储的正确率达到100%。     

A Speech Prosthesis Employing a Speech Synthesizer?Vowel Discrimination from Perioral Muscle Activities and Vowel Production

A speech prosthesis has been developed based on the following idea. When a handicapped person such as a laryngectomee tries to speak in vain, the movements of the mouth, tongue, etc., are elicited. By detecting the movements, what he or she is trying to say can be determined. Then a speech synthesizer is driven to produce a voice of good quality.     

基于临界频带的子带滤波语音增强

根据人耳的听觉感知特性,提出了一种基于子带滤波的优化语音增强方法。基于临界频带设计滤波器将输入信号分成若干子带,依据估计出的每个子带的短时信噪比来对相应子带的时域信号逐帧进行独立的自适应处理后再合成。语音增强性能评估结果表明,有效地去除背景噪声的同时还抑制了音乐噪声,减少了语音的听觉失真,提高了增强语音的可懂度。     

On some performance characteristics for a TASI system

The performance of a Time Assignment Speech Interpolation (TASI) system is usually characterized by the fraction of the freeze-out talkspurts of talkers. In the present investigation, we analyze the process of information transmission in a TASI system in details and introduce a new performance characteristic of such systems. Explicit analytical expressions are obtained under the assumption that input subscribers are continuously occupied, talkspurts and silence intervals are exponentially distributed.     

一种基于语音和噪声统计信息改进相位估计的语音增强算法

为改善低信噪比环境下语音的质量,论文提出了一种改进相位估计的语音增强算法。算法首先根据语音和噪声频谱的统计模型的对称性得到用先验信噪比倒数形式表示的噪声频谱估计值,然后通过分析低信噪比条件下（0dB）相位估计对于幅度估计的重要性,利用噪声频谱估计值估计每一个频点的相位修正值,并给出了一种优化的先验信噪比估计算法,得到一种新的语音增强算法。由仿真实验给出的客观测试和非正式听音测试表明:该算法处理后取得了较好的效果,在抑制低信噪比语音增强所产生的音乐噪声的前提下,相比未改进相位估计的算法处理后的信号,语音失真度更小,语音质量有明显提高。      

PLC-1: A TASI System for Small Trunk Groups

The PLC-1 Private Line Voice Concentrator is introduced. This equipment uses the well-known TASI (time assignment speech interpolation) technique to reduce the number of transmission facilities required to transmit a number of trunks by a factor of up to two. In addition to reducing transmission facility requirements, the PLC-1 provides a very high level of management and diagnostic tools to the telecommunications manager.     

A Prototype Switching System Employing a Matching-Based Language

This paper describes an experimental switching system employing a nonprocedural language. It presents a study of the appropriate architecture for implementing the language, and examines the system's performance characteristics. Furthermore, it identifies the critical part of the architecture, and provides two strategies for throughput improvement and indicates their effects.     

Delay Versus TASI Advantage in a Packet Voice Multiplexer

In this correspondence we complement, by means of an analytical model, an earlier simulation study on tradeoff between delay and TASI advantage in a packetized speech multiplexer.     

基于DSP的语音录放、存储及传输系统

在分析PC机ISA总线结构和PnP协议规范的基础上，结合DSP芯片自身的特点，通过扩展DSP的中断和DMA功能，实现了DSP对网卡和声卡的直接操作，从而设计了一套基于DSP的高性能语音录放、存储及传输系统。     

On a Speech Multiple System Implementation for Speech Synthesis

This paper proposes a voice synthesizer to convert a single speech to multiple speeches. Pitch is an important voice characteristic of speech parameter and it is produced by the periodic vibration of the vocal-cords; the parameter most sensitive for human’s auditory sense. So if you change the pitch of the voice with several different scaling rates, you can produce several different voices at the same time with a voice. The Multiple-Speech Synthesizer will be used in diverse practical applications. The cheer synthesizer that makes group voice from a single voice would be such an example. You could also use the system for a troll toy, birthday song tracks, effect sounds in movies and plays, protection systems of houses, etc. Furthermore, the synthesizer could be used to imitate the voice of famous entertainers or cartoon characters, such as the Mask-man.

Performance of a Multiple HAP System Employing Multiple Polarization

In this paper we address the potential gain of using compact MIMO antenna array configurations in conjunction with HAP (High Altitude Platforms) diversity techniques in order to increase the data rates in HAP communication systems. We will also investigate the effects of spatial correlation and mutual coupling between the separate antenna elements on system performance. Simulation results show that although the capacity is degraded by correlation and mutual coupling, we still achieve significant capacity gain compared to the single HAP case. In addition, we evaluate the performance of the system for different separation angles between HAPs, and determine the optimal separation angle that maximizes the total capacity of the system.     

大容量数字语音时间记录仪的实现

介绍了自行研制的一种基于DSP和CF卡的大容量双通道数字语音时间记录仪。阐述了系统的硬、软件设计及其功能与工作过程，给出了实际测试指标，最后介绍了该方案在铁路调度系统中的应用。     

一种云存储系统数据去重方案的性能分析

在数据去重的同时确保数据安全性是当前云存储系统亟待解决的挑战性问题,设计安全数据去重方案的目标是在确保存储空间利用率和数据安全性的前提下有效抵御内部和外部恶意攻击。文章对基于PoW(所有权证明)的安全云存储数据去重方案进行简要介绍,从PoW安全性、数据隐私性、攻击抵抗性和前后向安全等方面验证方案的安全性,并分析方案的密钥生成效率、数据上传效率和所有权管理能力。结果表明,在计算开销和通信开销方面,该方案具有比传统方案较为明显的优势。     

一种新的语音编码系统的设计

现有的大部分语音编码系统硬件框架复杂,软件设计难度较大.为实现低成本方案,设计了一种以SPCE061A语音单片机为基础的编码系统,压缩技术采用了基于正交滤波器的子带编码算法.系统硬件结构简洁、软件设计方便高效,具有较高的性价比.通过实验,对比了原始语音信号和重建信号的时域波形和回放质量,表明本系统具有一定的实用价值.     

Interference Analysis of an in-Building Wireless Communication System Employing Vertical Frequency Reuse

The reception quality of an interference limited indoor wireless communication system employing vertical frequency reuse is analysed. Outage probability expressions are used to measure the level of cochannel interference. The received signals are assumed to suffer the effects of Rayleigh fading and log-normal shadowing and multiple cochannel interferers are included in the analysis. The expected reception qualities are studied in a range of buildings and the results show that typically a vertical reuse distance of 3 floors will not isolate cochannel floors sufficiently to allow reliable reception. For one building analysed, even a vertical reuse distance of five floors is not likely to result in sufficiently low levels of cochannel interference. When multiple interferers are considered, it is not a straight forward task to determine the number of cochannel interferers that contribute significant interference since this may depend on the propagation conditions in a particular building.     

Delay Analysis of TASI with Random Fluctuations in the Number of Voice Calls

In this paper we consider a system where the talkspurts from several voice conversations are buffered and multiplexed over the same transmission channels. A simple mathematical model is given for the case where the number of voice calls may statistically come and go. This model is then used to numerically analyze the system and make comparisons with previously presented results.