卡耐基.梅隆大學(xué)的牛發(fā)寫的關(guān)于孤立點(diǎn)和數(shù)據(jù)清洗的文章，全英文，2003年完成，Probabilistic NOISE Identification and Data Cleaning，Real world data is never as perfect as we would like it to be and can often suffer from corruptions that may impact interpretations of the data, models created from the data, and decisions made based on the data. One approach to this problem is to identify and remove records that contain corruptions. Unfortunately, if only certain fields in a record have been corrupted then usable, uncorrupted data will be lost. In this paper we present LENS, an approach for identifying corrupted fields and using the remaining noncorrupted fields for subsequent modeling and analysis.
標(biāo)簽： 大學(xué) 數(shù)據(jù)
上傳時(shí)間： 2017-08-29
上傳用戶：thinode
This is an analog signal communication simulator, usign frequency modulation. It is designed in MATLAB-Simulink. The communications channel beetween the transmitter and the reciever is supposed to be affected by additive white Gaussian NOISE.
標(biāo)簽： communication modulation frequency simulator
上傳時(shí)間： 2013-12-25
上傳用戶：王楚楚
The TAS3204 is a highly-integrated audio system-on-chip (SOC) consisting of a fully-programmable, 48-bit digital audio processor, a 3:1 stereo analog input MUX, four ADCs, four DACs, and other analog functionality. The TAS3204 is programmable with the graphical PurePath Studio? suite of DSP code development software. PurePath Studio is a highly intuitive, drag-and-drop environment that minimizes software development effort while allowing the end user to utilize the power and flexibility of the TAS3204’s digital audio processing core. TAS3204 processing capability includes speaker equalization and crossover, volume/bass/treble control, signal mixing/MUXing/splitting, delay compensation, dynamic range compression, and many other basic audio functions. Audio functions such as matrix decoding, stereo widening, surround sound virtualization and psychoacoustic bass boost are also available with either third-party or TI royalty-free algorithms. The TAS3204 contains a custom-designed, fully-programmable 135-MHz, 48-bit digital audio processor. A 76-bit accumulator ensures that the high precision necessary for quality digital audio is maintained during arithmetic operations. Four differential 102 dB DNR ADCs and four differential 105 dB DNR DACs ensure that high quality audio is maintained through the whole signal chain as well as increasing robustness against NOISE sources such as TDMA interference. The TAS3204 is composed of eight functional blocks: Clocking System Digital Audio Interface Analog Audio Interface Power supply Clocks, digital PLL I2C control interface 8051 MCUcontroller Audio DSP – digital audio processing 特性 Digital Audio Processor Fully Programmable With the Graphical, Drag-and-Drop PurePath Studio? Software Development Environment 135-MHz Operation 48-Bit Data Path With 76-Bit Accumulator Hardware Single-Cycle Multiplier (28 × 48)
上傳時(shí)間： 2016-05-06
上傳用戶：fagong
設(shè)計(jì)中使用的信號(hào)為 信息信號(hào)： signal=sin(2*pi*sl*n*T) 高頻噪聲： NOISE =0.5*sin(2*pi*ns1*n*T) 混合信號(hào)： x=(signal+NOISE) 其中sl=1000Hz，ns1=4500Hz，T=1/10000。混合信號(hào)波形為濾波器輸入信號(hào)波形，信息信號(hào)波形為輸出信號(hào)波形，濾波器的效果為濾除兩個(gè)高頻噪聲。
上傳時(shí)間： 2016-05-08
上傳用戶：梅浩梅浩
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(shí)間： 2016-06-12
上傳用戶：halias
The AP2406 is a 1.5Mhz constant frequency, slope compensated current mode PWM step-down converter. The device integrates a main switch and a synchronous rectifier for high efficiency without an external Schottky diode. It is ideal for powering portable equipment that runs from a single cell lithium-Ion (Li+) battery. The AP2406 can supply 600mA of load current from a 2.5V to 5.5V input voltage. The output voltage can be regulated as low as 0.6V. The AP2406 can also run at 100% duty cycle for low dropout operation, extending battery life in portable system. Idle mode operation at light loads provides very low output ripple voltage for NOISE sensitive applications. The AP2406 is offered in a low profile (1mm) 5-pin, thin SOT package, and is available in an adjustable version and fixed output voltage of 1.2V, 1.5V and 1.8V
標(biāo)簽： 2406 AP 技術(shù)手冊(cè)
上傳時(shí)間： 2017-02-23
上傳用戶：w124141
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(shí)間： 2018-04-08
上傳用戶：kilohorse
%========================開始提取加噪信號(hào)的各類特征值================================ for n=1:1:50; m=n*Ns; x=(n-1)*Ns; for i=x+1:m; %提取加噪信號(hào)'signal_with_NOISE=y+NOISE'的前256個(gè)元素，抽取50次 y0(i)=signal_with_NOISE(i); end Y=fft(y0); %對(duì)調(diào)制信號(hào)進(jìn)行快速傅里葉算法（離散） y1=hilbert(y0) ; %調(diào)制信號(hào)實(shí)部的解析式 factor=0; %開始求零中心歸一化瞬時(shí)幅度譜密度的最大值gamma_max for i=x+1:m; factor=factor+y0(i); end ms=factor/(m-x); an_i=y0./ms; acn_i=an_i-1; end gamma_max=max(fft(acn_i.*acn_i))/Ns
標(biāo)簽： matlab 神經(jīng)網(wǎng)絡(luò)算法 通信信號(hào) 調(diào)制識(shí)別
上傳時(shí)間： 2020-04-07
上傳用戶：如拷貝般復(fù)制
%========================開始提取加噪信號(hào)的各類特征值================================ for n=1:1:50; m=n*Ns; x=(n-1)*Ns; for i=x+1:m; %提取加噪信號(hào)'signal_with_NOISE=y+NOISE'的前256個(gè)元素，抽取50次 y0(i)=signal_with_NOISE(i); end Y=fft(y0); %對(duì)調(diào)制信號(hào)進(jìn)行快速傅里葉算法（離散） y1=hilbert(y0) ; %調(diào)制信號(hào)實(shí)部的解析式 factor=0; %開始求零中心歸一化瞬時(shí)幅度譜密度的最大值gamma_max for i=x+1:m; factor=factor+y0(i); end ms=factor/(m-x); an_i=y0./ms; acn_i=an_i-1; end gamma_max=max(fft(acn_i.*acn_i))/Ns
標(biāo)簽： matlab 神經(jīng)網(wǎng)絡(luò)算法 通信信號(hào) 調(diào)制識(shí)別
上傳時(shí)間： 2020-04-07
上傳用戶：如拷貝般復(fù)制
Without conceding a blemish in the first edition, I think I had best come clean and admit that I embarked on a second edition largely to adopt a more geometric approach to the detection of signals in white Gaussian NOISE. Equally rigorous, yet more intuitive, this approach is not only student-friendly, but also extends more easily to the detection problem with random parameters and to the radar problem
標(biāo)簽： Communication Foundation Digital in
上傳時(shí)間： 2020-05-26
上傳用戶：shancjb
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