LTE下行链路的自适应MIMO技术方案

本文设计了LTE 下行链路在空间复用与传输分集技术之间自适应切换的多天线传输方案,实现了系统容量和传输质量的良好折中。系统仿真结果表明,自适应切换的多天线方案能够有效提高系统的吞吐量和覆盖范围。     

一种RFID密集天线切换电路的实现

为了实现UHF RFID密集天线切换,需要在切换电路之间传输控制信号、电源和900 MHz射频信号3种信号,采用传统做法需要3股电缆分别传输3种信号。切换电路利用MCU、晶体管、电阻、电容等简单器件的组合,可完成3种信号的合成和分离工作。通过MCU盘查天线切换系统的接入天线情况和天线切换控制。由此得到了一种较简单的电路,实现了一根同轴电缆传输3种信号,使整个系统连接简洁可靠。     

浅谈数字音频接口技术和D/A转换器

１数字音频接口ＡＥＳ／ＥＢＵ标准ＡＥＳ／ＥＢＵ标准定义了单线向传输数字音频数据的串行通信接口。遵照它的协议，各种数字音频设备如ＣＤ机、ＤＡＴ、ＤＡＣ等可以相互交换数字音频数据、定时信息、控制信息等，它分为专业用和民用两部分。１．１标准的演变过程专业用途的标准和民用用途的标准非常相似，它们由各种组织发布。它们的差异如表１所示。早期内部串行数字音频总线格式是ＳＯＮＹ公司在其ＰＣＡＩＦＩ数字音频处理机中应用的，这种连线方式包括二条信号线：ＤＡＴＡ，ＢＣＫ，ＬＲＣＫ。ＤＡＴＡ是交替传输的左右声道数据。Ｂ…     

浅谈数字音频接口技术和D／A转换器

１数字音频接口ＡＥＳ／ＥＢＵ标准ＡＥＳ／ＥＢＵ标准定义了单线单向传输数字音频数据的串行通信接口。遵照它的协议，各种数字音频设备如ＣＤ机、ＤＡＴ、ＤＡＣ等可以相互交换数字音频数据、定时信息、控制信息等，它分为专业用途和民用用途两部分。１－１标准的演变过程专业用途的标准和民用用途的标准非常相似，它们由各种组织发布。它们的差异如表１所示：早期内部串行数字音频总线格式是ＳＯＮＹ公司在其ＰＣＭＦ１数字音频处理机中应用的，这种连线方式包括三条信号线：ＤＡＴＡ、ＢＣＫ、ＬＲＣＫ。ＤＡＴＡ是交替传输的左右声道…     

Ethersound网络数字音频技术及应用

1引言随着数字网络技术的发展,一直从事网络数字音频应用研究的科技领先公司,在现有技术基础上不断推出更多更新的技术与产品。这些新技术新产品的出现,显示着网络化数字音频已经逐渐成为音频传输/切换/分配技术的方向,而率先倡导在专业音频的安装当中应用网络数字音频解决方案的PeakAudio和Digi-gram是当前全球网络化数字专业音频传输和交换的先驱者。所以无论是演出租赁公司还是大型工程项目的专业音频承包和安装者,现今的网络数字音频技术都能带来全新的竞争优势。2网络数字音频技术数字音频的核心实际上是数字音频压缩技术。MPEG音频…     

高性能的CMOS A／D和D／A转换器

ＡＤ９２６０Ａ／Ｄ（模拟数字）转换器和ＡＤ９７７４Ｄ／Ａ（数字模拟）转换器，具有极高的分辨率，很高的采样速率，大的信噪比，低的谐波失真，宽的无假频动态范围（ＳＦＤＲ）和低的互调干扰，以及低功耗等特性。其在数字音频和低频工业具有新的应用，比如，可以在一条电话线上以同一频率同时双向传输数字信号，又台几乎可以从天线直接接收主号，进行数字处理。     

信道差错对多发射天线选择性能的影响

对于多发单收天线(MISO)通信系统,考虑发射端信道状态信息的差错(ECSI).研究正交空时分组码的MISO系统的多天线选择,提出了一种CSI差错的选择性信道序统计特性求解方法.在独立的平坦瑞利衰落信道和多进制相移键控(MPSK)调制下,推导了一种较准确的系统比特误码率(BER)切诺夫上界解析式.最后系统BER性能上界的数值结果和仿真结果研究表明:多发射天线选择技术能极大地提高系统的传输质量,能有效地抵抗ECSI的影响.     

基于DAB及DAB系列系统的单频网切换方案设计与仿真

《30 MHz~3000 MHz地面数字音频广播系统技术规范》即通常所说的DAB(数字音频广播)中国标准虽然是针对于音频的,但是考虑到基带部分是信源透明的,所以完全可以作为手机电视的传输网络。文中在兼容DAB标准的情况下提出了一种单频网切换的方案,以实现服务的平滑过渡。通过对ETSI协议的相关补充并采用二级检索的机制保证了接收机在地域切换时信号的连续性,最后通过软件仿真的方法从视觉直观上验证了该方法的效果,从而确定了该方案的可行性。     

数字音频信号的传输技术

数字音频信号的应用具有一定的广泛性与普遍性,尤其是广播电视和录音设备,本文结合实际应用对AES/EBU数字音频信号的接口标准、传输技术进行分析,并分析数字信号传输的各种优点,包括抗干扰与再生修正等,最后介绍了以太网的CobraNet技术传输数字音频信号.     

一种基于HVS、模糊聚类和遗传算法的视频差错掩盖方法

当视频在传输过程中发生不可纠正的差错时,就要用差错掩盖的方法对丢失或差错的块进行掩盖。本文提出了一种基于模糊聚类和遗传算法的视频差错掩盖算法,首先,在确定适应度函数时,采用了一种适应人眼视觉系统(HVS: Human Visual System)的视频评价函数作为目标函数,其次,通过模糊聚类的方法同时综合多特征量得到差错块的相似块得到初始种群,然后利用GA的三个步骤:复制＼遗传＼交叉进行迭代地运算,直到最后得到满足条件的差错块的掩盖块。实验结果表明,其主客观效果比传统方法和基于一般GA算法的方法好。     

Frequency-Domain Stochastic Error Concealment for Wireless Audio Applications

Real time digital audio delivery over Wireless Local Area Networks (WLANs) represents an attractive, flexible and cost effective framework for realizing high-quality, multichannel home audio applications. However, the unreliable nature of WLANs IP link frequently imposes significant playback quality degradation, due to delay or permanent loss of a number of transmitted digital audio packets. In this paper, a novel packet error concealment technique is presented, based on the spectral reconstruction of the statistical equivalent of a previously successfully received audio data packet. It is shown that the proposed data reconstruction scheme outperforms previously published error concealment strategies, in both terms of objective and perceptual criteria.     

数字化会议系统的分析与设计

采用计算机或单片机为控制中心的数字化会议系统,实现对发言人员次序和摄像机的控制。使用单片机、A／D、D／A等主要器件,利用软件的方法控制与会者的发言．自动分配地址、语音溢出控制和防止自激等措施,实现会议系统的话音采集、传输和处理。与模拟系统相比,控制方便、系统容量大、话音质量好,适合在大型会议等场合使用。     

Joint encoding and decoding methods for digital audio broadcasting of multiple programs

We introduce new methods for increasing the performance of multiprogram digital audio broadcast systems, e.g., satellite digital audio broadcasting. Joint multiprogram encoding is an attractive possibility for parallel broadcasting of a large number of programs. Joint coding extended over multiple audio frames in time give further improvements. The benefits of this kind of statistical multiplexing yield improved audio quality and/or higher capacity in terms of number of programs. We describe the new Joint Multiple Program Encoding Technique in the context of the perceptual audio coding (PAC) type of algorithms. We also describe methods for multi-program transmission including Equal Error Protection (EEP) as well as Unequal Error Protection (UEP) and improved error concealment for multiple program transmission. Some of the techniques described in this paper, are currently being used in satellite digital audio broadcasting in the United States.     

An integrated error correction and detection system for digitalaudio broadcasting

Hybrid in-band on-channel digital audio broadcasting systems deliver digital audio signals in such a way that is backward compatible with existing analog FM transmission. We present a channel error correction and detection system that is well-suited for use with audio source coders, such as the so-called perceptual audio coder (PAC), that have error concealment/mitigation capabilities. Such error mitigation is quite beneficial for high quality audio signals. The proposed system involves an outer cyclic redundancy check (CRC) code that is concatenated with an inner convolutional code. The outer CRC code is used for error detection, providing flags to trigger the error mitigation routines of the audio decoder. The inner convolutional code consists of so-called complementary punctured-pair convolutional codes, which are specifically tailored to combat the unique adjacent channel interference characteristics of the FM band. We introduce a novel decoding method based on the so-called list Viterbi algorithm (LVA). This LVA-based decoding method, which may be viewed as a type of joint or integrated error correction and detection, exploits the concatenated structure of the channel code to provide enhanced decoding performance relative to decoding methods based on the conventional Viterbi algorithm (VA). We also present results of informal listening tests and other simulations on the Gaussian channel. These results include the preferred length of the outer CRC code for 96-kb/s audio coding and demonstrate that LVA-based decoding can significantly reduce the error flag rate relative to conventional VA-based decoding, resulting in dramatically improved decoded audio quality. Finally, we propose a number of methods for screening undetected errors in the audio domain     

An integrated temporal error concealment for H.264/AVC based on spatial evaluation criteria

Owing to error-prone transmission networks, the compressed video bit stream is prone to packet loss in the transmission channel. This loss causes serious distortion and the distortion will propagate to successive frames, especially in highly compressed video coding standard. Therefore, it is very important to efficiently enhance the restored result. In this paper, an integrated temporal error concealment technique for H.264/AVC is proposed. The technique could effectively restore the corrupted data by adaptively integrating error concealment approaches with the adaptive weight-based switching algorithm. The integrated mechanism is based on spatial evaluation criteria, judged by boundary distortion estimation and texture intensity. Experimental results show that the technique could effectively enhance the performance of error concealment.     

A power supply error correction method for single-ended digital audio class D amplifiers

In single-ended digital audio class D amplifiers (CDAs), the errors caused by power supply noise in the power stages degrade the output performance seriously. In this article, a novel power supply error correction method is proposed. This method introduces the power supply noise of the power stage into the digital signal processing block and builds a power supply error corrector between the interpolation filter and the uniform-sampling pulse width modulation (UPWM) lineariser to pre-correct the power supply error in the single-ended digital audio CDA. The theoretical analysis and implementation of the method are also presented. To verify the effectiveness of the method, a two-channel single-ended digital audio CDA with different power supply error correction methods is designed, simulated, implemented and tested. The simulation and test results obtained show that the method can greatly reduce the error caused by the power supply noise with low hardware cost, and that the CDA with the proposed method can achieve a total harmonic distortion + noise (THD + N) of 0.058% for a –3 dBFS, 1 kHz input when a 55 V linear unregulated direct current (DC) power supply (with the –51 dBFS, 100 Hz power supply noise) is used in the power stages.     

Improving the Reception Performance of Legacy T‐DMB/DAB Receivers in a Single‐Frequency Network with Delay Diversity

This paper describes a simple delay diversity technique for terrestrial digital multimedia broadcasting (T‐DMB) and digital audio broadcasting in a single‐frequency network (SFN). For the diversity technique, a delay diversity scheme is adopted. In the delay diversity scheme, a non‐delayed signal is transmitted in the first antenna, and delayed versions of the signal are transmitted in each additional antenna. For an SFN environment with multiple transmitters, delay diversity can be executed by controlling the emission times of the transmitters. This SFN delay diversity scheme does not require any hardware changes in either the transmitter or receiver, and perfect backward compatibility can be acquired. To evaluate the performance improvement, laboratory tests are executed with various types of commercial T‐DMB receivers as well as a measurement receiver. The improvement in the bit error rate performance is evaluated using a measurement receiver, and an improvement of the threshold of visibility value is evaluated for commercial receivers. Test results show that the T‐DMB system can obtain diversity gain using the described technique.     

基于功率D/A转换的音频功率放大器

设计了一种基于A/D和D/A相互转换的音频功率放大器.A/D转换后的数字音频信号经电平匹配和隔离驱动后,控制功率D/A转换电路进行音频还原和功率放大.当转换位数足够时,能基本不失真地还原音频信号.对功率D/A转换输出的阶梯波进行逐级分析,得出开关器件工作频率、器件通态损耗和开关损耗的计算式.利用多级自举方法,减少了驱动电源数目.实验结果表明,这是一种效率较高的音频功率放大器.     

MF digital AM station antennas

It is very important having the proper antenna, specially in the MF AM band where achieving the necessary bandwidth in the antenna impedance is a difficult task. This problem is critical in the lower frequencies were the antenna matching to the transmission line generally is very sharp and the best match is obtained only at the carrier frequency with an appreciable power reflection in the lateral frequency bands. This problem is not very important in the classical AM transmissions were the maximum transmitted power is located at the carrier frequency and only a fraction in the upper and lower lateral bands. Of course this produces some distortion in the AM transmission but in this case the quality of the audio is not really of high fidelity, like in an FM transmission, due to a lot of factors, one of them the lack of the full audio spectrum. This problem can be corrected with high fidelity audio transmitters and specially with digital transmission in order to achieve CD quality audio and here the transmitting antenna plays an important role. In this paper MF AM antenna systems are analyzed not only from the input impedance point of view, but with consideration of all the factors in order to determine the best system in bandwidth and radiation properties in different parts of the standard AM band. Cylindrical, type A, and Cantilever classical monopoles and the modern dipole type antenna systems are compared, to provide the criteria for choosing an optimum antenna for the future digital AM service. Examples of measured field strength as a function of distance in a flat region are presented in order to show the interesting MF AM possibilities for a digital service     

高精度大动态范围音频数字增益控制器设计

设计了一种用于音频领域的高精度大动态范围对数型数字增益控制器,实现了在音频范围内的电压到电压对数型衰减控制,衰减范围为0~-94.5 dB,单位衰减量为-1.5 dB。增益控制字为8位信号DIN[7∶0],内部使用18位R-2R结构DAC实现精确衰减控制,使用对数译码逻辑减少输入控制字位数。设计了具有低噪声特性的轨到轨运算放大器,等效输入噪声仅为10.7 nV/Hz@1 kHz。仿真结果表明,设计的对数型数字增益控制器具有工作电压低、动态范围大、衰减精度高、等效输入噪声低等优点。