检索结果-内蒙古大学图书馆

Performance of dsp-assisted gray-coded DP-16-QAM Nyquist-DWDM super-channel transceiver for high-speed long-haul optical interconnects

引用

PHOTONIC NETWORK COMMUNICATIONS 2023年第1期45卷 37-48页

作者： Kakati, Dhiman Sonkar, Ramesh Kumar Indian Inst Technol Guwahati Dept Elect & Elect Engn Gauhati 781039 India

In this paper, a 64-channel Nyquist-DWDM system has been proposed for spectral-efficient long-haul optical interconnects. Dual-polarization 16-QAM has been used at each transmitter, and the homodyne detection technique decodes the transmitted symbols at the receiver. Gray code used in the transmitters shows an improved OSNR tolerance, receiver sensitivity, and transmission distance in comparison to the differential code. Erbium-doped fiber amplifiers compensate for attenuation at regular intervals in the multi-span transmission line. Digital signal processing techniques with advanced algorithms compensate for the signal impairments after demodulation. The transceiver performance has been analyzed through comprehensive numerical methods for the 25 GHz and 50 GHz DWDM grids. The system performance has been optimized considering various parameters such as laser power, linewidth, crosstalk, OSNR tolerance, and receiver sensitivity. The transceiver achieves 7.68 Tbps data rate over a transmission distance of 3100 km and 2650 km for the considered DWDM grids, respectively. The gray-coded system possesses an OSNR tolerance of 16.4 dB and the overall DWDM system spectral efficiency of 4.71 b/s/Hz at its maximum transmission distance. This super-channel transceiver shall be helpful in high-capacity long-haul transmission and backbone support to the existing network infrastructure.

关键词： Nyquist-WDM Gray code dsp algorithms Coherent detection Super-channel transceiver

来源：评论

学校读者我要写书评

暂无评论

Real-time Audio Effects with dsp algorithms and DirectSound 6

Real-time Audio Effects with DSP Algorithms and DirectSound

引用

6th International Conference on Electronics, Computers and Artificial Intelligence (ECAI)

作者： Anghelescu, Petre Angelescu, Stefan Ionita, Silviu Univ Pitesti Dept Elect Commun & Comp Pitesti Romania

ISBN: (纸本)9781479954797

This paper is focused on the software implementation of dsp oriented algorithms for different audio effects, like Echo, 3-Tap Echo, Vibrato, Tremolo and Chorus. The main contribution of this research consists in a software application which is able to process in real time the audio signals received from the electric guitar using PC sound card. In addition, the application also includes a digital filter class used for implementation of a low pass filter as well as other filter types, a tone generator and the options for storing the sound samples in a wav file. The application is written in C# language which uses DirectSound library in order to process the sound samples on a low level.

关键词： Digital audio effects dsp algorithms DirectSound Electric guitar

来源：评论

学校读者我要写书评

暂无评论

A novel design of low-cost hearing aid devices using an efficient lifting filter bank with a modified variable filter

引用

EXPERT REVIEW OF MEDICAL DEVICES 2022年第12期19卷 991-999页

作者： Subbulakshmi, N. Manimegalai, R. Rajakumar, G. Kumar, T. Ananth Kosuri, Umadevi Indian Inst Technol Comp Sci & Engn Gauhati Assam India PSG Inst Technol & Appl Res Comp Sci & Engn Coimbatore Tamil Nadu India Francis Xavier Engn Coll Elect & Engn Tirunelveli Tamil Nadu India IFET Coll Engn Comp Sci & Engn Villupuram Tamil Nadu India Gokaraju Rangaraju Inst Engn & Technol Comp Sci & Engn Hyderebad India IFET Coll Engn Villupuram India

ObjectivesIn recent years, audio signal processing begins its outbreak on a global scale, which has altered the global market and has rapidly extended out as a fundamental design in various industries. Lifting based method is frequently implemented in many applications such as image processing and audio processing because it has some advantages that faster implementation with minimum computational *** this proposed work, a Novel Lifting based Filter Bank (NLFB) is proposed using a modified variable filter aimed at digital hearing aid devices. This filter bank design has some mandatory constraints, such as hardware complexity, power consumption, and delay. The proposed method is designed with four lifting steps such as split, predict, update and merge to make the perfect reconstruction in analysis and decomposition in synthesis bank. The performance analysis of the proposed method is discussed in this *** proposed method consumes less power, up to 45mW, and a minimum delay between 85ns and 91.1ns when compared to traditional methods. ConclusionThe proposed design output consumes 32 % of minimum hardware components, 12% of low power compared to interpolated filter bank and 6% of delay is reduced using Modified Variable Filter (MVFB).

关键词： Finite impulse response filter bank dsp algorithms signal processing hearing aids digital filters

来源：评论

学校读者我要写书评

暂无评论

FPGA Implementation of RNS Adder Based MAC Unit in Ternary Value Logic Domain for Signal Processing Algorithm and its Performance Analysis

FPGA Implementation of RNS Adder Based MAC Unit in Ternary V...

引用

IEEE International Conference on Electron Device Kolkata Conference (EDKCON)

作者： Ghosh, Aniruddha Sinha, Amitabha Calcutta Inst Technol ECE Dept Howrah WB India Birbhum Inst Engn & Technol Birbhum WB India

ISBN: (纸本)9781538664155

Digital signal processing (dsp) based applications are designed using various types of dsp algorithms which are computationally intensive. So, dsp-based applications widely utilize Multiply-Accumulate (MAC) operation for accomplishing speed. In contrast with the binary number system, Residue Number Systems (RNS) is considered to be more prominent because of their abilities of carrying out carry-free arithmetic operations like addition, subtraction. Ternary value logic (TVL) offers several advantages like reduced chip area as well as overall delay, over conventional binary number system. Designing superior adder and multiplier have become the major concern for implementing high performance signal processing applications. To improve the performance of MAC unit, a new architecture is proposed in this paper. In this paper, MAC unit is implemented using ternary multiplier and RNS adder in TVL domain. The major bottleneck of TVL to RNS conversion and vice versa has introduced huge complexity which leads to decreased efficiency of performance due to large conversion time. The performance of RNS based system can be enhanced by choosing relative prime moduli set as improper selection of moduli will affect system speed, dynamic range and hardware complexity. Proposed MAC unit is mapped on field programmable gate array (FPGA) for analysis its performance.

关键词： Ternary Value Logic (TVL) Trit Ternary Gates Ternary Residue Number Systems (TRNS) Ternary Multiplier TRNS Adder Multiply and Accumulate Unit (MAC) FPGA dsp algorithms

来源：评论

学校读者我要写书评

暂无评论

Human footstep detection using seismic sensors 7

Human footstep detection using seismic sensors

引用

7th International Conference on Electronics, Computers and Artificial Intelligence (ECAI)

作者： Anghelescu, Petre Iana, Gabriel Vasile Tramandan, Ionut Univ Pitesti Dept Elect Commun & Comp Pitesti Romania SC Mira Telecom SA Mira Telecom Software Dept Bucharest Romania

ISBN: (纸本)9781467366472

This paper is focused on the hardware and software implementation of dsp algorithms for footstep detection using seismic signal. The main contribution of this research consists in a hardware design and in a software application that process the signals received from the seismic sensor of type SM-24 using PC sound card. The signal samples received from the sensor are stored in a. wav file. The main application is written in C# programming language and uses Matlab library (it was created a DLL library that was used in C#) in order to process the seismic signals on a low level.

关键词： Seismic footstep detection dsp algorithms Kurtosis Cadence Feature extraction

来源：评论

学校读者我要写书评

暂无评论

Tri-Axial Telecoil Hearing Aid for Improved Connection to Public Induction Loops 37

Tri-Axial Telecoil Hearing Aid for Improved Connection to Pu...

引用

37th Annual International Conference of the IEEE-Engineering-in-Medicine-and-Biology-Society (EMBC)

作者： Riehle, Timothy H. Knuesel, Robert J. Lichter, Patrick A. Panescu, Dorin Koronis Biomed Technol Inc Maple Grove MN 55369 USA Adv Cardiac Therapeut Santa Clara CA USA

ISBN: (纸本)9781424492701

Telecoils in hearing aids serve a dual purpose: to enhance telephone conversation and enable hearing aids to serve as a wireless interface for public audio broadcasts. When broadcasting audio signals, the signal is transmitted to the telecoil sensor (a magnetic field sensor located in the hearing aid) via magnetic energy from an induction wire loop located near the listener. This induction loop can be a small assembly located in the handset of the telephone or a large wire loop within a public venue like a theatre. Current hearing aids detect the magnetic signal using a single telecoil. If the telecoil is not aligned with the magnetic field, the strength of the detected signal is diminished. Unfortunately, public induction loops and telephone handsets seldom share a common alignment to the hearing aid, leading to sub-optimal performance by one or both in hearing aids available today. In this research, a prototype behind-the-ear (BTE) hearing aid with tri-axial telecoils was developed and dsp algorithms to process and combine the three signals were developed. The resulting hearing aid was evaluated in a human trial and provided better performance than a similar quality conventional telecoil hearing aid.

关键词： audio signal processing hearing aids telephone sets dsp algorithms audio signals induction wire loop listener magnetic energy magnetic field sensor magnetic signal prototype behind-the-ear hearing aid public audio broadcasts public induction loops telecoil sensor telephone conversation telephone handset triaxial telecoil hearing aid wireless interface Auditory system Channel estimation Digital signal processing Hearing aids Prototypes Signal to noise ratio Speech Hearing Aids Telephone sets magnetic signals radio interface Magnetic sensors magnetic energy audio signals Listeners audio signal processing signal to noise ratios Digital signal processing Auditory System Induction coils channel estimation PROTOTYPE

来源：评论

学校读者我要写书评

暂无评论

Vendor-Interoperable Elastic Optical Interfaces: Standards, Experiments, and Challenges

引用

JOURNAL OF OPTICAL COMMUNICATIONS AND NETWORKING 2015年第12期7卷 B184-B193页

作者： Gunkel, Matthias Mattheus, Arnold Wissel, Felix Napoli, Antonio Pedro, Joao Costa, Nelson Rahman, Talha Meloni, Gianluca Fresi, Francesco Cugini, Filippo Sambo, Nicola Bohn, Marc Deutsch Telekom FMED D-64295 Darmstadt Germany Coriant ATG D-81541 Munich Germany Coriant Portugal P-2720093 Amadora Portugal IST Inst Telecomun Lisbon Portugal Eindhoven Univ Technol COBRA Inst NL-5600 MB Eindhoven Netherlands CNIT Pisa Italy

This paper aims to review the status of standardization activities on the black link (BL) and corresponding alien wavelength concepts. It discusses experimental work on filterless optical networks conducted within a dedicated Deutsche Telekom project. The general prospects and existing challenges concerning elastic extensions of the BL are also assessed. Furthermore, we present relevant work on control and management plane interoperability comprising generalized multi-protocol label switching and transport software defined networking aspects. In the second part of the article, we report on the latest dual-vendor experiments on data plane interoperability in terms of digital signal processing technology for next generation 1.28 Tb/s PM-16QAM super-channels. Finally, we present a network reachability analysis based on the widely used Gaussian noise model in the context of data plane interoperability. This analysis estimates the network-wide impact of the single-vendor versus the dual-vendor approach.

关键词： Data plane interoperability dsp algorithms Elastic black link Elastic optical networks FEC Transport SDN

来源：评论

学校读者我要写书评

暂无评论

Fast Fixed-Point Optimization of dsp algorithms

Fast Fixed-Point Optimization of DSP Algorithms

引用

18th IEEE/IFIP International Conference on VLSI and System-on-Chip

作者： Caffarena, Gabriel Carreras, Carlos Lopez, Juan A. Fernandez, Angel Univ San Pablo CEU Madrid 28668 Spain Univ Politecn Madrid E-28040 Madrid Spain

ISBN: (纸本)9781424464708

In this paper, a fixed-point quantization noise estimator aiming at Digital Signal Processing (dsp) algorithms is presented. The estimator enables significant reduction in the computation time required to perform complex word-length optimizations while providing a high accuracy. Affine Arithmetic (AA) is used to provide a Signal-to-Quantization Noise-Ratio (SQNR) estimation for differentiable non-linear algorithms with and without feedbacks. The estimation is based on the parameterization of the statistical properties of the noise at the output of fixed-point algorithms. This parameterization allows to relate the fixed-point formats of the signals with the output noise distribution by means of fast matrix operations. Thus, a fast estimation is achieved and the word-length optimization computation time is significantly reduced. The estimator is tested using a subset of non-linear algorithms such as vector operations, power computation IIR filter, adaptive filters, and channel equalizers. The Word-length optimization computation time is boosted by three orders of magnitude while keeping the average estimation error down to 6% for most cases.

关键词： dsp algorithms SQNR estimation affine arithmetic digital signal processing algorithms digital signal processing chips fast fixed-point optimization fast matrix operations fixed point arithmetic fixed-point quantization noise estimator matrix algebra noise statistical property nonlinear algorithms nonlinear estimation optimisation signal-to-quantization noise-ratio estimation word-length optimization computation time

来源：评论

学校读者我要写书评

暂无评论

A Survey of Filter Bank algorithms for Biomedical Applications 4

A Survey of Filter Bank Algorithms for Biomedical Applicatio...

引用

4th International Conference on Computer Communication and Informatics (ICCCI)

作者： Subbulakshmi, N. Manimegalai, R. Sri Ramakrishna Engn Coll Dept ECE Coimbatore Tamil Nadu India Pk Coll Engn & Technol Dept CSE Coimbatore Tamil Nadu India

ISBN: (纸本)9781479923526

The Digital signal processing algorithms are promising techniques, which are used to alleviate the filter bank designs in various applications. Filter bank is an enabling technique for numerous capabilities such as speech coding, Noise reduction, Sub band coding and auditory compensation. The design procedure mainly concentrates on the core part of the system which is the type of the filter. However, it imposes several solutions to many problems high computation complexity, area and power consumption are most critical concern. In digital hearing aid application, the filter bank improves the sound ability for hearing-impaired people. The scope of this work is to give an overview of the filter bank algorithms under various traits and the challenges that they face, along with the current state-of-the-art. To enhance the feasibility of the design by applying the characteristics of the filter banks for suitable applications. This paper covers wide range of issues in the design of filter bank. The contribution of this paper is threefold. First, we show the functional role of filter bank. Second, the classifications of filter bank algorithms for the hearing instruments. Third, merits, demerits and further design challenges of the filter banks are discussed.

关键词： component Filter bank Digital hearing aid dsp algorithms

来源：评论

学校读者我要写书评

暂无评论

A Domain-specific Language to Facilitate Software Defined Radio Parallel Executable Patterns Deployment on Heterogeneous Architectures

A Domain-specific Language to Facilitate Software Defined Ra...

引用

33rd IEEE International Performance, Computing, and Communications Conference (IPCCC)

作者： Mohapi, Lerato J. Winberg, Simon Inggs, Michael Univ Cape Town Radar Remote Sensing Grp Dept Elect Engn ZA-7925 Cape Town South Africa

ISBN: (纸本)9781479975754

In this paper, we present a domain-specific language, referred to as OptiSDR, that matches high level digital signal processing (dsp) routines for software defined radio (SDR) to their generic parallel executable patterns targeted to heterogeneous computing architectures (HCAs). These HCAs includes a combination of hybrid GPU-CPU and dsp-FPGA architectures that are programmed using different programming paradigms such as C/C++, CUDA, OpenCL, and/or VHDL. OptiSDR presents an intuitive single high-level source code and near specification-level approach for optimization and facilitation of HCAs. OptiSDR uses an optimized embedded domain-specific language (DSL) compiler framework called Delite. Our focus is on the programming language expressiveness for parallel programming and optimization of typical dsp algorithms for deployment on SDR HCAs. We demonstrate the capability of OptiSDR to express the solution to the issues of parallel dsp low-level implementation complexities in the closest way to the original parallel programming of SDR systems. This paper will achieve these by focusing on three generic parallel executable patterns suitable for dsp routines such as cross-correlation, convolution in FIR filter based Hilbert transformers, and fast Fourier transforms for spectral analysis. This paper concludes with a performance analysis using dsp algorithms that tests automatically generated code against hand-crafted solutions.

关键词： FIR filters Hilbert transforms convolution fast Fourier transforms parallel programming program compilers programming languages signal processing software radio spectral analysis telecommunication computing DSL compiler framework dsp algorithms dsp routines dsp-FPGA architectures Delite FIR filter OptiSDR SDR HCA cross-correlation digital signal processing domain-specific language heterogeneous computing architectures high-level source code hybrid GPU-CPU parallel executable patterns deployment programming language programming paradigms software defined radio Computer architecture DSL Digital signal processing Finite impulse response filters Graphics processing units Programming

来源：评论

学校读者我要写书评

暂无评论

建议与咨询留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

时间限定

文献类型

馆藏选择

核心期刊

语言

文献类型

帮助

文字说明：

检索规则说明：

检索范例：

分类表

所选分类

限定检索结果

文献类型

馆藏范围

日期分布

学科分类号

主题

机构

作者

语言

请选择保存的检索档案：

请选择收藏分类：

通借通还

建议与咨询 留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

时间限定

文献类型

馆藏选择

核心期刊

语言

文献类型

帮助

文字说明：

检索规则说明：

检索范例：

分类表

所选分类

限定检索结果

文献类型

馆藏范围

日期分布

学科分类号

主题

机构

作者

语言

请选择保存的检索档案： 新增检索档案 确定 取消

请选择收藏分类： 新增自定义分类 确定 取消

通借通还

建议与咨询留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

请选择保存的检索档案：

请选择收藏分类：