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)
上传时间： 2016-05-06
上传用户：fagong
Color appearance models aim to extend basic colorimetry to the level of specifying the perceived color of stimuli in a wide variety of viewing conditions
标签： appearance Color model
上传时间： 2016-05-21
上传用户：wisewater
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
msp430The LDC1312 and LDC1314 are 2- and 4-channel, 1• Easy-to-use – minimal configuration required 12-bit inductance to digital converters (LDCs) for • Measure up to 4 sensors with one IC inductive sensing solutions. With multiple channels • Multiple channels support environmental and and support for remote sensing, the LDC1312 and aging compensation LDC1314 enable the performance and reliability benefits of inductive sensing to be realized at minimal• Multi-channel remote sensing provides lowest cost and power. The products are easy to use, onlysystem cost requiring that the sensor frequency be within 1 kHz • Pin-compatible medium and high-resolution and 10 MHz to begin sensing. The wide 1 kHz to 10 options MHz sensor frequency range also enables use of very small PCB coils, further reducing sensing– LDC1312/4: 2/4-ch 12-bit LDC solution cost and size.– LDC1612/4: 2/4-ch 28
上传时间： 2016-07-22
上传用户：tongmoonsky
HTML5 从入门到精通主要讲述HTML5 视频、音乐、canvas、veb存储、input类型等。
标签： HTML5
上传时间： 2016-10-24
上传用户：nhwswjsd
#include <malloc.h> #include <stdio.h> #include <stdlib.h> #include <string.h> #define NULL 0 #define MaxSize 30 typedef struct athletestruct /*运动员*/ { char name[20]; int score; /*分数*/ int range; /**/ int item; /*项目*/ }ATH; typedef struct schoolstruct /*学校*/ { int count; /*编号*/ int serial; /**/ int menscore; /*男选手分数*/ int womenscore; /*女选手分数*/ int totalscore; /*总分*/ ATH athlete[MaxSize]; /**/ struct schoolstruct *next; }SCH; int nsc,msp,wsp; int ntsp; int i,j; int overgame; int serial,range; int n; SCH *head,*pfirst,*psecond; int *phead=NULL,*pafirst=NULL,*pasecond=NULL; void create(); void input () { char answer; head = (SCH *)malloc(sizeof(SCH)); /**/ head->next = NULL; pfirst = head; answer = 'y'; while ( answer == 'y' ) { Is_Game_DoMain: printf("\nGET Top 5 when odd\nGET Top 3 when even"); printf("\n输入运动项目序号 (x<=%d):",ntsp); scanf("%d",pafirst); overgame = *pafirst; if ( pafirst != phead ) { for ( pasecond = phead ; pasecond < pafirst ; pasecond ++ ) { if ( overgame == *pasecond ) { printf("\n这个项目已经存在请选择其他的数字\n"); goto Is_Game_DoMain; } } } pafirst = pafirst + 1; if ( overgame > ntsp ) { printf("\n项目不存在"); printf("\n请重新输入"); goto Is_Game_DoMain; } switch ( overgame%2 ) { case 0: n = 3;break; case 1: n = 5;break; } for ( i = 1 ; i <= n ; i++ ) { Is_Serial_DoMain: printf("\n输入序号 of the NO.%d (0<x<=%d): ",i,nsc); scanf("%d",&serial); if ( serial > nsc ) { printf("\n超过学校数目,请重新输入"); goto Is_Serial_DoMain; } if ( head->next == NULL ) { create(); } psecond = head->next ; while ( psecond != NULL ) { if ( psecond->serial == serial ) { pfirst = psecond; pfirst->count = pfirst->count + 1; goto Store_Data; } else { psecond = psecond->next; } } create(); Store_Data: pfirst->athlete[pfirst->count].item = overgame; pfirst->athlete[pfirst->count].range = i; pfirst->serial = serial; printf("Input name:) : "); scanf("%s",pfirst->athlete[pfirst->count].name); } printf("\n继续输入运动项目(y&n)？"); answer = getchar(); printf("\n"); } } void calculate() /**/ { pfirst = head->next; while ( pfirst->next != NULL ) { for (i=1;i<=pfirst->count;i++) { if ( pfirst->athlete[i].item % 2 == 0 ) { switch (pfirst->athlete[i].range) { case 1:pfirst->athlete[i].score = 5;break; case 2:pfirst->athlete[i].score = 3;break; case 3:pfirst->athlete[i].score = 2;break; } } else { switch (pfirst->athlete[i].range) { case 1:pfirst->athlete[i].score = 7;break; case 2:pfirst->athlete[i].score = 5;break; case 3:pfirst->athlete[i].score = 3;break; case 4:pfirst->athlete[i].score = 2;break; case 5:pfirst->athlete[i].score = 1;break; } } if ( pfirst->athlete[i].item <=msp ) { pfirst->menscore = pfirst->menscore + pfirst->athlete[i].score; } else { pfirst->womenscore = pfirst->womenscore + pfirst->athlete[i].score; } } pfirst->totalscore = pfirst->menscore + pfirst->womenscore; pfirst = pfirst->next; } } void output() { pfirst = head->next; psecond = head->next; while ( pfirst->next != NULL ) { // clrscr(); printf("\n第%d号学校的结果成绩:",pfirst->serial); printf("\n\n项目的数目\t学校的名字\t分数"); for (i=1;i<=ntsp;i++) { for (j=1;j<=pfirst->count;j++) { if ( pfirst->athlete[j].item == i ) { printf("\n %d\t\t\t\t\t\t%s\n %d",i,pfirst->athlete[j].name,pfirst->athlete[j].score);break; } } } printf("\n\n\n\t\t\t\t\t\t按任意建 进入下一页"); getchar(); pfirst = pfirst->next; } // clrscr(); printf("\n运动会结果:\n\n学校编号\t男运动员成绩\t女运动员成绩\t总分"); pfirst = head->next; while ( pfirst->next != NULL ) { printf("\n %d\t\t %d\t\t %d\t\t %d",pfirst->serial,pfirst->menscore,pfirst->womenscore,pfirst->totalscore); pfirst = pfirst->next; } printf("\n\n\n\t\t\t\t\t\t\t按任意建结束"); getchar(); } void create() { pfirst = (struct schoolstruct *)malloc(sizeof(struct schoolstruct)); pfirst->next = head->next ; head->next = pfirst ; pfirst->count = 1; pfirst->menscore = 0; pfirst->womenscore = 0; pfirst->totalscore = 0; } void Save() {FILE *fp; if((fp = fopen("school.dat","wb"))==NULL) {printf("can't open school.dat\n"); fclose(fp); return; } fwrite(pfirst,sizeof(SCH),10,fp); fclose(fp); printf("文件已经成功保存\n"); } void main() { system("cls"); printf("\n\t\t\t 运动会分数统计\n"); printf("输入学校数目 (x>= 5):"); scanf("%d",&nsc); printf("输入男选手的项目(x<=20):"); scanf("%d",&msp); printf("输入女选手项目(<=20):"); scanf("%d",&wsp); ntsp = msp + wsp; phead = (int *)calloc(ntsp,sizeof(int)); pafirst = phead; pasecond = phead; input(); calculate(); output(); Save(); }
标签： 源代码
上传时间： 2016-12-28
上传用户：150501
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
上传时间： 2017-02-23
上传用户：w124141
J1为Line input 5V，两个104电容为滤波电容，引脚分别接地，主要滤掉高频纹波，防止自激振荡；47UF和100UF为滤波电容，主要滤掉低频纹波；R1作限流作用以保护稳压二极管D2，当输入电压和输出负载电流发生变化时R1通过本身压降的变化，来调节稳压二极管D2的工作电流，从而起到稳压作作用。
上传时间： 2017-07-29
上传用户：18826226223
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
BP神经网络matlab源程序代码 %% 该代码为 BP 神经网络的预测代码 % 清空环境变量 clc clear % %% 网络结构建立 % 读取数据 load data input output input=data(;,1;2;3;4;5); output=data(;,5); %节点个数 %训练数据和预测数据 %选连样本输入输出数据归一化 %构建网络 net=newff(inputn,outputn,hiddennum); %网络进化参数 %网络训练 % 预测数据归一化
上传时间： 2018-04-10
上传用户：45808330
