虫虫首页| 资源下载| 资源专辑| 精品软件
登录| 注册

MIXED-Signal

  • dspc语言程序实现iir滤波器

    设计中使用的信号为 信息信号: signal=sin(2*pi*sl*n*T) 高频噪声: noise =0.5*sin(2*pi*ns1*n*T) 混合信号: x=(signal+noise) 其中sl=1000Hz,ns1=4500Hz,T=1/10000。混合信号波形为滤波器输入信号波形,信息信号波形为输出信号波形,滤波器的效果为滤除两个高频噪声。

    标签: dspc iir 语言程序 滤波器

    上传时间: 2016-05-08

    上传用户:梅浩梅浩

  • advanced signal

    jiaocao yongyu tongxingongcheng xinhao xuexi 

    标签: advanced signal

    上传时间: 2016-06-07

    上传用户:sz869

  • Microphone Arrays : A Tutorial

    This report presents a tutorial of fundamental array processing and beamforming theory relevant to microphone array speech processing. A microphone array consists of multiple microphones placed at different spatial locations. Built upon a knowledge of sound propagation principles, the multiple inputs can be manipulated to enhance or attenuate signals emanating from particular directions. In this way, microphone arrays provide a means of enhancing a desired signal in the presence of corrupting noise sources. Moreover, this enhancement is based purely on knowledge of the source location, and so microphone array techniques are applicable to a wide variety of noise types. Microphone arrays have great potential in practical applications of speech processing, due to their ability to provide both noise robustness and hands-free signal acquisition.

    标签: Microphone array Tutorial Array Signal Processing

    上传时间: 2016-06-12

    上传用户:halias

  • 传感器网络中基于到达时间差有效的凸松弛方法的稳健定位

    We consider the problem of target localization by a network of passive sensors. When an unknown target emits an acoustic or a radio signal, its position can be localized with multiple sensors using the time difference of arrival (TDOA) information. In this paper, we consider the maximum likelihood formulation of this target localization problem and provide efficient convex relaxations for this nonconvex optimization problem.We also propose a formulation for robust target localization in the presence of sensor location errors. Two Cramer-Rao bounds are derived corresponding to situations with and without sensor node location errors. Simulation results confirm the efficiency and superior performance of the convex relaxation approach as compared to the existing least squares based approach when large sensor node location errors are present.

    标签: 传感器网络

    上传时间: 2016-11-27

    上传用户:xxmluo

  • TMS320VC5416-DSK-example

    A simple example of audio signal processing on TMS320VC5416 USB DSK board. Main source is contained in tone.c file, memory configuration - tonecfg.cmd. Folder docs/ contains useful docmentation on board, its components and libraries. The example's configuration is based on example "tone" from Code Composer Studio's 3.1 example for 5416 DSK.

    标签: DSK-example 5416 TMS 320 VC

    上传时间: 2017-09-29

    上传用户:wang1104014663

  • 基于频率插值的4.0kbps 语音编码器的性能和设计(英文)

    The 4.0 kbit/s speech codec described in this paper is based on a Frequency Domain Interpolative (FDI) coding technique, which belongs to the class of prototype waveform Interpolation (PWI) coding techniques. The codec also has an integrated voice activity detector (VAD) and a noise reduction capability. The input signal is subjected to LPC analysis and the prediction residual is separated into a slowly evolving waveform (SEW) and a rapidly evolving waveform (REW) components. The SEW magnitude component is quantized using a hierarchical predictive vector quantization approach. The REW magnitude is quantized using a gain and a sub-band based shape. SEW and REW phases are derived at the decoder using a phase model, based on a transmitted measure of voice periodicity. The spectral (LSP) parameters are quantized using a combination of scalar and vector quantizers. The 4.0 kbits/s coder has an algorithmic delay of 60 ms and an estimated floating point complexity of 21.5 MIPS. The performance of this coder has been evaluated using in-house MOS tests under various conditions such as background noise. channel errors, self-tandem. and DTX mode of operation, and has been shown to be statistically equivalent to ITU-T (3.729 8 kbps codec across all conditions tested.

    标签: frequency-domain interpolation performance Design kbit_s speech coder based and of

    上传时间: 2018-04-08

    上传用户:kilohorse

  • bpsk调制的仿真

    用matlab实现bpsk调制 用randi函数产生10000个0、1随机数,对原信号signal进行判断,       若为0,则生成调制信息1,若为1,则生成调制信息-1,从而实现BPSK调制。

    标签: bpsk 调制 仿真

    上传时间: 2018-04-25

    上传用户:yyqqyy

  • qpsk仿真

    利用randi函数产生10000个0、1随机数,对原信号signal进行分类每两个相邻随机数作为新信息SIGNAL并进行判断,若为0,则生成调制信息1+i,若为1,则生成调制信息-1+i,若为2,则生成调制信息-1-i,若为3,则生成调制信息1-i,从而实现QPSK调制

    标签: qpsk 仿真

    上传时间: 2018-04-25

    上传用户:yyqqyy

  • BTS50_datasheet

    The BTS5016SDA is a one channel high-side power switch in PG-TO252-5-11 package providing embedded protective functions. The power transistor is built by a N-channel vertical power MOSFET with charge pump. The design is based on Smart SIPMOS chip on chip technology. The BTS5016SDA has a current controlled input and offers a diagnostic feedback with load current sense and a defined fault signal in case of overload operation, overtemperature shutdown and/or short circuit shutdown.

    标签: datasheet BTS 50

    上传时间: 2019-03-27

    上传用户:guaixiaolong

  • 短时过零率和短时能量

    % Computation of ST-ZCR and STE of a speech signal. % % Functions required: zerocross, sgn, winconv. % % Author: Nabin Sharma % Date: 2009/03/15 [x,Fs] = wavread('so.wav'); % word is: so x = x.'; N = length(x); % signal length n = 0:N-1; ts = n*(1/Fs); % time for signal % define the window wintype = 'rectwin'; winlen = 201; winamp = [0.5,1]*(1/winlen);

    标签: 短时过零率和短时能量

    上传时间: 2019-09-23

    上传用户:minwenji