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1.
在研究IEEE 802.16标准关于VoIP传输机制的基础上,指出语音流的静默特性严重制约了VoIP的容量.通过分析VoIP业务流的统计特性得出了在统计复用情况下VoIP的容量极限.提出了一种基于统计复用的带宽调度方法及其实现算法,以克服由于语音静默带来的语音速率波动.仿真结果表明,该调度方法与UGS(unsolicited grant service)调度相比能够显著提升VoIP的容量,同时使BS和SS之间的VoIP丢包率和传输时延保持在0.02%和5.1ms以下.  相似文献   

2.
龙恳  万溢  刘畅  田霖 《高技术通讯》2016,(3):255-263
分析了通过基站休眠降低无线通信网络能效的集中式算法和分布式算法的原理和性能,在此基础上,针对集中式休眠算法随着网络规模的增大其计算复杂度将会异常巨大的问题,提出了一种面向大规模通信网络的集中式分簇算法。该算法首先在时域上运用多目标均衡优化对休眠时间段进行划分,随后在空域上对基站进行合理的分簇,最后通过粒子群优化算法进行了休眠组合的确定。仿真结果表明,该算法的计算复杂度要低于其他集中式算法,并且性能衰减可以忽略不计,整体的休眠部署更加合理。  相似文献   

3.
邱晶  冯文江 《高技术通讯》2008,18(2):137-141
针对WCDMA系统软切换的位置选择分集发射(SSDT)方式,提出了一种自适应导频功率调整和主小区选择算法。采用这种算法,各基站能够根据各自小区负载和覆盖情况动态调整其导频功率,进行相邻小区间的负载均衡,另外,以系统效用最大化为目标,为每个用户选择最佳主小区。仿真结果表明,该算法能以可接受的计算复杂度换取系统性能的提高。  相似文献   

4.
彭坦  龚晨  李晔  洪侃  崔慧娟  唐昆 《高技术通讯》2008,18(5):452-457
为了提高在高误码率窄带无线信道下的合成语音质量,提出了一种信源信道联合编解码保护的语音编码抗误码算法。该算法在编码端利用编码后的冗余度进行BCH编码和奇偶校验以保护对语音合成质量影响较大的参数;在解码端对清浊音参数采用分支判决和改进的最大后验概率算法进行恢复,在浊音帧对线谱对(LSP)参数进行基于信源信道联合特性的线谱对参数差错后处理,在清音帧采用BCH解码和前向替代。该算法在不消耗任何额外带宽且无算法延时的条件下可以显著提高语音编码抗信道误码能力和恶劣信道条件下的合成语音质量。仿真实验显示,在较高信道误码率下平均谱失真降低了25%~36.1%,平均意见得分(MOS)提高了12.33%。  相似文献   

5.
This paper investigates the resource optimization problem for a multi-cell massive multiple-input multiple-output (MIMO) network in which each base station (BS) is equipped with a large number of antennas and each base station (BS) adapts the number of antennas to the daily load profile (DLP). This paper takes into consideration user location distribution (ULD) variation and evaluates its impact on the energy efficiency of load adaptive massive MIMO system. ULD variation is modeled by dividing the cell into two coverage areas with different user densities: boundary focused (BF) and center focused (CF) ULD. All cells are assumed identical in terms of BS configurations, cell loading, and ULD variation and each BS is modeled as an M/G/m/m state dependent queue that can serve a maximum number of users at the peak load. Together with energy efficiency (EE) we analyzed deployment and spectrum efficiency in our adaptive massive MIMO system by evaluating the impact of cell size, available bandwidth, output power level of the BS, and maximum output power of the power amplifier (PA) at different cell loading. We also analyzed average energy consumption on an hourly basis per BS for the model proposed for data traffic in Europe and also the model proposed for business, residential, street, and highway areas.  相似文献   

6.
Wang  F. Liu  Z. Song  X. 《Communications, IET》2009,3(2):257-267
In wireless communications, a sleep mode is commonly used to save power for mobile stations (MS). When there is no traffic to transmit, an MS periodically switches to sleep mode. Obviously, the performance of a power-saving mechanism depends on its sleep mode scheduling algorithm and the traffic characteristics of the user. In real systems, the power-saving mechanism of IEEE802.11 WLAN uses a constant sleep interval, and the IEEE802.16e WMAN adopts one with truncated exponentially extending sleep intervals denoted by PS-16, which contains constant sleep intervals as special cases. The two mechanisms are compared, resulting in the, finding that in the case of Poisson traffic, they have the same performance; whereas in the case of non-Poisson traffic PS-16 has better performance. For non-Poisson traffic, the performance of PS-16 lacks a closed form expression, which makes its design challenging. The authors propose to approximate the idle durations of an MS by hyper-exponentials, based on which an online sleep mode scheduling algorithm is developed. Numerical examples are provided to illustrate the efficiency of the proposed algorithm.  相似文献   

7.
1Time-scale representation of voiced speech is applied to voice quality analysis, by introducing the Line of Maximum Amplitude (LoMA) method. This representation takes advantage of the tree patterns observed for voiced speech periods in the time-scale domain. For each period, the optimal LoMA is computed by linking amplitude maxima at each scale of a wavelet transform, using a dynamic programming algorithm. A time-scale analysis of the linear acoustic model of speech production shows several interesting properties. The LoMA points to the glottal closure instants. The LoMA phase delay is linked to the voice open quotient. The cumulated amplitude along the LoMA is related to voicing amplitude. The LoMA spectral centre of gravity is an indication of voice spectral tilt. Following these theoretical considerations, experimental results are reported. Comparative evaluation demonstrates that the LoMA is an effective method for the detection of Glottal Closure Instants (GCI). The effectiveness of LoMA analysis for open quotient, amplitude and spectral tilt estimations is also discussed with the help of some examples.  相似文献   

8.
9.
This paper investigates the effects of temporal clipping on perceived speech quality. Temporal clipping usually results from voice activity detection (VAD), or line echo canceller's nonlinear processor, and the clipped speech portions are replaced by comfort noise. A nonintrusive algorithm is proposed to predict speech quality based on the clipping statistics. Mean opinion score (MOS) is used as a metric for speech quality and is measured by perceptual evaluation of speech quality (PESQ). The impacts of speech frame size and noise spectrum on the algorithm are also investigated. The results show that the proposed algorithm can efficiently predict the speech quality. The correlation coefficient between the prediction and the measurement is about 0.975, and the root mean square error for the prediction is 0.20 MOS. The algorithm can be used as an integral part of a general speech quality assessment scheme in voice over Internet protocol (VoIP).  相似文献   

10.
姜卫东  雷辉  郭勇 《声学技术》2014,33(2):176-179
针对水声传感器网络的簇间路由选择问题,提出了一种基于前向网关的低时延能耗均衡路由算法,该算法采用最优方向角原则和能耗均衡原则选择中继簇头和中继网关,以减小长延迟和高能耗对水声通信的影响。仿真结果表明该算法在网络平均能耗、端到端时延和网络生命周期等方面具有较好的性能。  相似文献   

11.
The IEEE 802.11n standard has provided prominent features that greatly contribute to ubiquitous wireless networks. Over the last ten years, voice over IP (VoIP) has become widespread around the globe owing to its low-cost or even free call rate. The combination of these technologies (VoIP and wireless) has become desirable and inevitable for organizations. However, VoIP faces a bandwidth utilization issue when working with 802.11 wireless networks. The bandwidth utilization is inefficient on the grounds that (i) 80 bytes of 802.11/RTP/UDP/IP header is appended to 10–730 bytes of VoIP payload and (ii) 765 µs waiting intervals follow each 802.11 VoIP frame. Without considering the quality requirements of a VoIP call, be including frame aggregation in the IEEE 802.11n standard has been suggested as a solution for the bandwidth utilization issue. Consequently, several aggregation methods have been proposed to handle the quality requirements of VoIP calls when carried over an IEEE 802.11n wireless network. In this survey, we analyze the existing aggregation methods of VoIP over the A-MSDU IEEE 802.11n wireless standard. The survey provides researchers with a detailed analysis of the bandwidth utilization issue concerning the A-MSDU 802.11n standard, discussion of the main approaches of frame aggregation methods and existing aggregation methods, elaboration of the impact of frame aggregation methods on network performance and VoIP call quality, and suggestion of new areas to be investigated in conjunction with frame aggregation. The survey contributes by offering guidelines to design an appropriate, reliable, and robust aggregation method of VoIP over 802.11n standard.  相似文献   

12.
浊音端点检测在语音处理中占有重要地位,在语音编解码、语音识别、语音增强处理中都需要用到端点检测技术。常规的以短时能量、过零率等作为判决特征参数的方法无法在低信噪比的系统中满足应用需求。本文以信号的共振峰和基音周期检测为基础检测浊音端点,算法首先提取语音信号的第一共振峰以及基音周期信息并以此为判决依据判断浊音的起点和终点。实验证明,这种方法在噪声环境中相对于传统的基于能量检测的或AMR_WB标准中的端点检测算法有更高的正确率。  相似文献   

13.
Push-to-talk (PTT) is a walkie-talkie like service which performs an efficient and instant voice over Internet protocol (VoIP) communication in mobile ad hoc network especially for certain circumstances, such as battle field and earthquake or disaster relief. The authors have designed and implemented the PTT mechanism in ad hoc VoIP network. The PTT server and user agent combined with the pseudo session initiation protocol (SIP) server in the implementation provide the PTT service without standalone SIP server support. The authors also conduct the experimental measurements, in terms of delay and packet loss, in the test-bed to demonstrate the realisation of PTT service in ad hoc VoIP network.  相似文献   

14.
Wireless Sensor Networks (WSNs) have hardware and software limitations and are deployed in hostile environments. The problem of energy consumption in WSNs has become a very important axis of research. To obtain good performance in terms of the network lifetime, several routing protocols have been proposed in the literature. Hierarchical routing is considered to be the most favorable approach in terms of energy efficiency. It is based on the concept parent-child hierarchy where the child nodes forward their messages to their parent, and then the parent node forwards them, directly or via other parent nodes, to the base station (sink). In this paper, we present a new Energy-Efficient clustering protocol for WSNs using an Objective Function and Random Search with Jumps (EEOFRSJ) in order to reduce sensor energy consumption. First, the objective function is used to find an optimal cluster formation taking into account the ratio of the mean Euclidean distance of the nodes to their associated cluster heads (CH) and their residual energy. Then, we find the best path to transmit data from the CHs nodes to the base station (BS) using a random search with jumps. We simulated our proposed approach compared with the Energy-Efficient in WSNs using Fuzzy C-Means clustering (EEFCM) protocol using Matlab Simulink. Simulation results have shown that our proposed protocol excels regarding energy consumption, resulting in network lifetime extension.  相似文献   

15.
首先介绍了一种可用于两跳中继网络的时分模式帧结构,然后基于该帧结构提出了一种资源分配算法.该算法包括两个步骤:第一个步骤是第一个时隙上的动态资源分配,第二个步骤是基站与中继站之间的最优化功率分配.两跳中继网络中下行存在三种链路:"基站-中继"链路、"基站-用户"链路和"中继-用户"链路,在系统资源有限的条件下,通过所提算法为这三种链路分配相应的带宽和功率,可以使得系统的吞吐量最大.仿真结果表明所提算法能够有效地提高系统的吞吐量.  相似文献   

16.
针对大规模MIMO系统用配置单精度模数转换器(ADC)来降低系统能耗会导致系统损失部分性能的问题,提出了一种莱斯信道下大规模MIMO系统的采用混合ADC的随机向量量化(RVQ)改进方案。该方案在频分双工模式下,首先在基站端采用高分辨率ADC和低分辨率ADC混合的接收方案处理信号,使接收的导频信号和有用信号具有较高的转换精度;系统对导频信号进行信道估计后,再对信道状态信息(CSI)进行RVQ处理,以此减小系统的反馈开销;最后运用最小均方误差(MMSE)信号检测算法减轻由量化误差引起的的用户间干扰,从而达到降低能耗并减小系统性能损失的目的。实验结果表明,这种改进的RVQ方案能在降低系统能耗的基础上有效减小系统容量损失,并使其和速率接近传统的高分辨率ADC接收方案。  相似文献   

17.
The Internet of Things (IoT) is gaining attention because of its broad applicability, especially by integrating smart devices for massive communication during sensing tasks. IoT-assisted Wireless Sensor Networks (WSN) are suitable for various applications like industrial monitoring, agriculture, and transportation. In this regard, routing is challenging to find an efficient path using smart devices for transmitting the packets towards big data repositories while ensuring efficient energy utilization. This paper presents the Robust Cluster Based Routing Protocol (RCBRP) to identify the routing paths where less energy is consumed to enhances the network lifespan. The scheme is presented in six phases to explore flow and communication. We propose the two algorithms: i) energy-efficient clustering and routing algorithm and ii) distance and energy consumption calculation algorithm. The scheme consumes less energy and balances the load by clustering the smart devices. Our work is validated through extensive simulation using Matlab. Results elucidate the dominance of the proposed scheme is compared to counterparts in terms of energy consumption, the number of packets received at BS and the number of active and dead nodes. In the future, we shall consider edge computing to analyze the performance of robust clustering.  相似文献   

18.
Transient ultrasonic waves in a fluid medium generated by a flat circular and a point-focused transducer of finite size are modeled by the distributed point source method (DPSM). DPSM is a Green's-function-based semi-analytical mesh-free technique which is modified here to incorporate the transient loading from a finite-sized acoustic transducer. Conventional DPSM solves acoustic problems in steady-state frequency domain. Here, DPSM is extended to the time domain without the fast Fourier transform (FFT) but using the Green's function in the time domain. This modified method is denoted t-DPSM. Harmonic point sources of DPSM are replaced by time-dependent point sources in t-DPSM. Generated t-DPSM results are compared with the finite element (FE) results for both focused and flat circular transducers. The developed method is used to solve the transient problem of wave scattering by an air bubble in a fluid as the bubble is moved horizontally or vertically from the focal point of the focused transducer. The received energy signal is compared for different eccentricities.  相似文献   

19.
A simple mechanism to prolong the life cycle of the network by balancing nodes’ energy consumption is to rotate the active dominating set (DS) through a set of legitimate DSs. This paper proposes a novel adaptive clustering algorithm named HREF (Highest Remaining Energy First). In the HREF algorithm, cluster formation is performed cyclically and each node can declare itself as a cluster head autonomously if it has the largest residual energy among all its adjacent nodes. The performance effectiveness of the HREF algorithm is investigated and compared to the D-WCDS (Disjoint Weakly Connected Dominating Set) algorithm. In this paper, we assume the network topology is fixed and does not require sensor mobility. This allows us to focus on the impact of clustering algorithms on communication between network nodes rather than with the base station. Simulation results show that in the D-WCDS algorithm energy depletion is more severe and the variance of the node residual energy is also much larger than that in the HREF algorithm. That is, nodes’ energy consumption in the HREF algorithm is in general more evenly distributed among all network nodes. This may be regarded as the main advantage of the HREF adaptive clustering algorithm.  相似文献   

20.
曹文婧  朱敏  武岩波 《声学技术》2017,36(5):473-478
船舶噪声是影响水下语音通信质量的主要因素。为了提高单边带语音信号的信噪比和语音质量,采用多通道自适应增强算法对语音信号进行研究。该算法运用自适应信号增强技术,具有运算量小、易实时实现、消噪效果显著等优点,并且能够有效利用不同换能器接收到的信息。从理论上对该算法进行了分析,并结合海试数据进行验证。结果表明,低信噪比、非平稳噪声环境下,该算法有效抑制了接收信号中的尖锐噪声,并且当系统收敛时,使其趋于白化,明显提高了系统的输出信噪比,显著改善了语音质量,在性能上明显优于传统的信号相干叠加算法。  相似文献   

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