共查询到17条相似文献,搜索用时 234 毫秒
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随着移动通信技术的快速发展,语音增强的研究及其实际应用成为数字化通信的一个重要的研究方向。在数字信号处理技术的支撑下,许多优秀的语音增强算法的实时实现成为了可能。谱减法是一种运算量相对较小,增强效果明显,并且容易实时实现的语音增强算法,但是其缺点就是残留有音乐噪声。针对传统谱减法,本语音增强系统采用了一种改进算法,就是... 相似文献
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基于E-Model的VoIP语音质量测量 总被引:1,自引:0,他引:1
基于E-Model的语音质量测量方法是一种客观测试方法,它克服了传统语音质量测试在数据网络测量中的不足。为了能够准确评估VoIP语音质量,在E-Model算法的基础之上,探讨了延时、噪声、回音、语音压缩等损伤因素对VoIP语音质量的影响。 相似文献
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双通道能量差后滤波语音增强算法在语音通信系统的噪声抑制技术中有较好的应用前景,然而其理论性能和局限性还未得到充分研究。为此,本文采用统计分析方法研究了双通道能量差后滤波语音增强算法的性能,分析了相干性、平滑因子及噪声估计误差对算法的影响。理论和仿真结果表明,噪声估计误差和平滑因子严重影响该算法的降噪性能。依据此分析结果,本文提出一种基于非平稳噪声估计和功率谱自适应平滑的双通道能量差后滤波算法。测试结果表明,本文提出的算法在不增加语音失真的前提下,能更有效地抑制非平稳噪声,段信噪比提高(SegSNRI)和语音质量感知评估(PESQ)等客观评价指标都表明本文的算法优于其它几种经典的后滤波算法。 相似文献
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VoIP是基于互联网协议下使用网络进行语音信号传递的技术,实现了传统通信网络和互联网的结合,近年来在通信领域得到了长足的发展和广泛的应用。本文主要对VoIP技术在民航VHF通信系统中的应用进行分析,并探讨了VoIP技术未来的发展方向,希望能给VoIP技术在民航通信网络中更为广泛的应用提供参考。 相似文献
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为了抑制小型语音通信设备中的方向性噪声干扰问题,提出了一种结合差分阵列与幅度谱减的双麦语音增强算法。该算法首先利用一阶差分阵列技术,对两麦克风采集到的带噪语音信号进行处理,得到语音通道信号和噪声通道信号。接着利用差分阵列处理后的两通道信号对语音通道信号的信噪比进行估计。最后利用幅度谱减法对语音通道信号中残留噪声进行消除。针对语音通道信号的信噪比估计,本文给出了两种新奇的计算方法。仿真实验表明,该算法有效的抑制了方向噪声,改善了语音的质量,去噪效果及语音质量均优于对比算法。 相似文献
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Aiming to introduce voice over IP networks and services in ways that satisfy the voice quality expectations of our customers, we have been conducting laboratory studies of how VoIP transmission affects voice quality while also carefully monitoring and managing several field implementations of VoIP. This article summarizes much of what we have learned in this work, and we hope it provides a useful progress report on the industry's evolution to VoIP. We review our data on the voice quality effects of packet loss, delay, speech coders, packet loss concealment algorithms, and the compression option of suppressing transmission during silence. Because the familiar problem of echo has emerged repeatedly in the VoIP environment, we review this issue in some detail. Packet loss and delay variation measurements made on private VoIP networks are reviewed, and the data here are encouraging. We finish by making our case that the network planning tool known as the E-model is currently an inexact predictor of VoIP network performance. 相似文献
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Voice communications over zigbee networks 总被引:3,自引:0,他引:3
Chonggang Wang Sohraby K. Jana R. Lusheng Ji Daneshmand M. 《Communications Magazine, IEEE》2008,46(1):121-127
This article provides an overview of ZigBee-enabled wireless networks and discusses the feasibility of supporting voice communications over ZigBee networks. We begin by providing an overview of the ZigBee technology followed by an evaluation of voice quality and performance over such an impoverished wireless channel. Two types of voice communications, namely full-duplex voice over IP (VoIP) and half-duplex push-to-talk (PTT) are considered. Voice quality of VoIP is measured using the R-factor [1] (a well known objective speech quality metric). The quality of PTT, however, is evaluated based on packet-loss rate, delay, and jitter. The simulation results demonstrate that a low-power, low-rate wireless sensor network can support a limited range of voice services. 相似文献
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Sung-Min Oh Sunghyun Cho Jae-Hyun Kim Jonghyung Kwun 《Communications Letters, IEEE》2008,12(5):374-376
This letter proposes an efficient uplink scheduling algorithm for voice over Internet protocol (VoIP) services with adaptive multi-rate (AMR) speech codec in IEEE 802.16e/m systems. The proposed scheduling algorithm adopts the random access scheme during silent-period to reduce the waste of uplink bandwidth considering the characteristics of AMR speech codec. The numerical results show that the proposed algorithm can increase the maximum supportable number of voice users by 26% compared to the conventional extended real-time polling service (ertPS). 相似文献
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The paper presents a speech coding algorithm for operation at 11025 samples/s. The coder provides improved speech quality and compatibility with the MS‐Windows multimedia environment. The coding algorithm has been developed by adapting the ITU G729 and enhancing it with some recent developments in the medium band coding. The coder operates over a band of frequencies ranging from 20 to 5400 Hz at a bit rate of 8.9 kbit/s. Application of this coder includes intranet VoIP, voice chatting, multimedia communications, and voice archiving. Copyright © 2001 John Wiley & Sons, Ltd. 相似文献
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VoIP以语音流为传输媒介,具有传输数据量大和应用广泛的优点。但VoIP系统也面临数据安全和隐私泄露的安全威胁。针对编码标准G.729固定码本搜索的非遍历特性和具有一定冗余性的特点,该文提出基于G.729语音编码非零脉冲位置信息的隐藏算法。该算法在固定码本搜索过程中,利用秘密信息控制码本的搜索过程,并在非零脉冲位置和秘密信息之间构建函数进行信息隐藏。在搜索过程中利用最不重要脉冲替换思想并采用最小化失真准则控制由秘密信息的嵌入带来的音质失真。实验结果表明:算法隐藏容量可达400 bit/s,算法具有良好的隐蔽性(PESQ平均值约为3.45)。 相似文献
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讨论了一种基于传统谱相减算法的改进方法。利用语音的短时平稳性,通过先验幅度比来连续更新噪声谱的估计,从而代替复杂的VAD(话音活性检测)。计算机仿真结果表明,这种改进方法有效抑制了噪声干扰,语音得到了增强,在极大地提高信噪比的同时,将残留的音乐噪声和语音失真保持在人耳听觉容忍的范围以内,从而较好的保持了语音自然度。 相似文献