第一章JSP簡介 例子1（效果如圖1.1所示） Example1_1.jsp： < @ page contentType="text/html charset=GB2312" > <HTML> <BODY BGCOLOR=cyan> <FONT Size=1> <P>這是一個簡單的JSP頁面 < int i, sum=0 for(i=1 i<=100 i++) { sum=sum+i } >
標(biāo)簽： contentType Example charset 2312
上傳時間： 2017-08-31
上傳用戶：zycidjl
RLE for text compression.
標(biāo)簽： compression text RLE for
上傳時間： 2014-11-23
上傳用戶：xcy122677
Commercially available active noise control headphones rely on fixed analog controllers to drive "anti-noise" loudspeakers. Our design uses an adaptive controller to optimally cancel unwanted acoustic noise. This headphone would be particularly useful for workers who operate or work near heavy machinery and engines because the noise is selectively eliminated. Desired sounds, such as speech and warning signals, are left to be heard clearly. The adaptive control algorithm is implemented on a Texas Instruments (TI™ ) 1 TMS320C30GEL digital signal processor (DSP), which drives a Sony CD550 headphone/microphone system. Our experiments indicate that adaptive noise control results in a dramatic improvement in performance over fixed noise control. This improvement is due to the availability of high-performance programmable DSPs and the self-optimizing and tracking capabilities of the adaptive controller in response to the surrounding noise.
標(biāo)簽： Commercially controllers headphones available
上傳時間： 2013-12-04
上傳用戶：dyctj
一共有三種方式來發(fā)送和接收SMS信息：Block Mode, Text Mode和PDU Mode。其中PDU Mode被所有手機(jī)支持，可以使用任何字符集，這也是手機(jī)默認(rèn)的編碼方式。其中又分7bit-160，8bit-140，16bit-70的方式，我們中文用16bit70的方式。
上傳時間： 2014-12-06
上傳用戶：bakdesec
Text file encryptor is a user friendly application to encrypt text messages
標(biāo)簽： application encryptor friendly messages
上傳時間： 2013-12-01
上傳用戶：sssl
This mambot adds to the external links in a content the text “ target="_blank"” . Thus, all the links pointing to external sites will be opened in a new window.
標(biāo)簽： the external content mambot
上傳時間： 2013-12-17
上傳用戶：zhaiyanzhong
Data distill from text file
標(biāo)簽： distill Data from file
上傳時間： 2013-12-31
上傳用戶：libinxny
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.
標(biāo)簽： Microphone array Tutorial Array Signal Processing
上傳時間： 2016-06-12
上傳用戶：halias
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.
標(biāo)簽： frequency-domain interpolation performance Design kbit_s speech coder based and of
上傳時間： 2018-04-08
上傳用戶：kilohorse
% 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);
標(biāo)簽： 短時過零率和短時能量
上傳時間： 2019-09-23
上傳用戶：minwenji
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