Description: 采样率转换代码，在数字信号处理过程中，实现不同采样率系统的数据传送。-sample rate conversion code in digital signal processing to achieve different sampling rate of data transmission. Platform: |
Size: 11619 |
Author:张承云 |
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Description: 采样率转换代码，在数字信号处理过程中，实现不同采样率系统的数据传送。-sample rate conversion code in digital signal processing to achieve different sampling rate of data transmission. Platform: |
Size: 11264 |
Author:张承云 |
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Description: 音频采样率变换程序，可以加到程序后处理以便调整音视频同步，同时也可以用作预处理，降低采样率。是当前最好的音频采样率转换程序，被mpeg使用-audio sampling rate conversion programs can be added after processing procedures so as to adjust the audio and video synchronization, also can be used for pretreatment, lower sampling rate. Is currently the best audio sampling rate conversion process, the use of mpeg Platform: |
Size: 131072 |
Author:李东 |
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Description: 录音和播放程序，可以自由改变录音的Sample rate，并存成WAV格式，在EVC4.0上开发。-Sound recording and playback process, are free to change the recording of the Sample rate, both into WAV format, developed in EVC4.0. Platform: |
Size: 118784 |
Author:凌峰 |
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Description: The file is a tutorial introduction to Decimation, one of the sample rate conversion processes. It contains detailed basic exposition of the process and a design example to introduce the beginner to the concept. Platform: |
Size: 32768 |
Author:Anchit |
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Description: The Zoom-FFT is a process where an input signal is mixed down to baseband and then decimated, prior to passing it into a standard FFT. The advantage is for example that if you have a sample rate of 10 MHz and require at least 10Hz resolution over a small frequency band (say 1 KHz) then you do not need a 1 Mega point FFT, just decimate by a factor of 4096 and use a 256 point FFT which is obviously quicker.
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Size: 5120 |
Author:张鹏程 |
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Description: 采样率转换程序，程序主体分为上采样与下采样-Sample rate conversion process, the program is divided into the main sampling and down-sampling Platform: |
Size: 1024 |
Author:Jack Bauer |
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Description: g.726语音编解码标准的使用，整个程序的功能如下：
1）读入原始的WAV语音文件（Audio sample rate:8khz,Audio sample size:8bit）
2) wav语音文件转为PCM语音文件
3）对PCM语音文件进行G.726压缩编码
4）对压缩后的文件进行解压缩转为PCM语音文件
5）把PCM语音文件转为WAV语音文件-g.726 speech coding standards for the use of the functions of the whole process is as follows: 1) read the original WAV audio files (Audio sample rate: 8khz, Audio sample size: 8bit) 2) wav audio file to PCM audio file 3) G.726 PCM voice files on the compression 4) the compressed files to PCM audio files extract 5), the PCM audio files to WAV audio files Platform: |
Size: 324608 |
Author:陈智铭 |
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Description: 本文介绍一种基于AD1674设计的接口电路,该电路具有查询、中断和DMA三种数据传输功能,同时采用8253定时脉冲或端口写两种A/D启动方式。其中DMA方式实现了数据的快速传输,而两种A/D启动方式将会使采样率的设定更加灵活。应用本文原理设计的可插入通用PC机的数据采集板已用于我们的高频多谱勒和到达角探测分析系统中,取得了满意的效果。这些设计方法和原理在其它实际数据采集系统的设计过程中,也会具有重要的参考价值。
-This paper describes a design based on the AD1674 interface circuit, the circuit has a query, three types of interrupt and DMA data transfer function, while by 8253 the port was timed pulse or two A/D start mode. In which DMA was achieved by the rapid transmission of data, while the two A/D start mode will set the sample rate more flexible. Application of this principle of universal design can be inserted into PC-data acquisition board has been used in our high-frequency Doppler, and angle of arrival detection analysis system with satisfactory results. These design methods and principles in other practical data acquisition system design process, will be an important reference value. Platform: |
Size: 59392 |
Author:wsh |
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Description: 用MATLAB实现采样率转换的程序，附有说明文档，包括内插和抽取，以及先内插再抽取的程序-Using MATLAB to achieve sample-rate conversion process, with documentation, including interpolation and decimation, and interpolation and then taken before the procedure Platform: |
Size: 53248 |
Author:xiaosolider |
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Description: Basic communication theory requires that a minimum sampling rate of twice the frequency of the signal to be sampled will result in an accurate representation of the original signal. Because the human voice does not exceed 4000 Hz, a sampling rate of 8000 samples per second, or once every 125 microseconds, is used.
The conversion process begins by analyzing each sample and converting it into an 8-bit word also called an octet. If there are 8 bits per sample and 8000 samples per second, then the product is 64,000 bits per second (64 kbps)—the digital voice channel.
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Size: 1114112 |
Author:Arsh |
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Description: 抽样定理演示程序，能清楚看到当抽样速率大于2fs时恢复模拟信号的过程和当抽样频率小于2fs时频率混叠的过程。同时还包括频率抽样过程的演示。非常详尽。通过此演示程序，你肯定能明白抽样定理这一通信领域基本的原理！-The sampling theorem demo program, you can clearly see that when the recovery process of the analog signal sampling rate greater than 2fs and frequency aliasing when the sampling frequency is less than 2fs process. Also includes a demonstration of the frequency of the sampling process. Very detailed. Through this demonstration program, you can certainly understand sampling theorem this basic principle in the field of communication! Platform: |
Size: 7168 |
Author:hardi |
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Description: 泊松过程的模拟。并通过对样本函数处理，统计去均值函数和方差函数。而且对泊松过程的速率进行估计。-Poisson process simulation. And processing of the sample function, the statistics to go mean function and variance function. And to estimate the rate of the Poisson process. Platform: |
Size: 1024 |
Author:红薯 |
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Description: 速率方程求解，的详细过程。还有范例,方便一步一步学习-Rate equations, the detailed process. There sample rate equations, the detailed process. There are examples Platform: |
Size: 69632 |
Author:海星 |
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Description: AdaBoost元算法属于boosting系统融合方法中最流行的一种，说白了就是一种串行训练并且最后加权累加的系统融合方法。
具体的流程是：每一个训练样例都赋予相同的权重，并且权重满足归一化，经过第一个分类器分类之后，
计算第一个分类器的权重alpha值，并且更新每一个训练样例的权重，然后再进行第二个分类器的训练，相同的方法.......
直到错误率为0或者达到指定的训练轮数，其中最后预测的标签计算是各系统*alpha的加权和，然后sign（预测值）。
可以看出，训练流程是串行的，并且训练样例的权重是一直在变化的，分错的样本的权重不断加大，正确的样本的权重不断减小。
AdaBoost元算法是boosting中流行的一种，还有其他的系统融合的方法，比如bagging方法以及随机森林。
对于非均衡样本的处理，一般可以通过欠抽样(undersampling)或者过抽样(oversampling)，欠抽样是削减样本的数目，
过抽样是重复的选取某些样本，最好的方法是两种进行结合的方法。
同时可以通过删除离决策边界比较远的样例。
-AdaBoost boosting systems dollar fusion algorithm is the most popular one, it plainly systems integration approach is a serial train and final weighted cumulative.
Specific process is: Each training example is given equal weight, and the weights satisfy normalization, after the first classifiers after
Calculating a first classifier weights alpha value for each sample and updates right weight training, and then the second classifier training, the same way .......
0, or until the specified error rate training rounds, wherein the label is the calculation of the final prediction system* alpha weighted and then sign (predicted value).
As can be seen, the training process is serial, and weight training examples is always changing, the right of the wrong sample weight continued to increase, the right to correct sample weight decreasing.
AdaBoost algorithm is an element, as well as other methods of boosting popular systems integration, such as bagging and random forest method.
For Platform: |
Size: 2048 |
Author:iihaozl |
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Description: 用Matlab编程，实现解决该问题的单样本训练BP网络，设置一个停止迭代的误差Emin和最大迭代次数。在调试过程中，通过不断调整隐层节点数，学习率η，找到收敛速度快且误差小的一组参数。产生均匀分布在区间[-4，4]的测试样本，输入建立的模型得到输出，与Hermit多项式的期望输出进行比较计算总误差（运行5次，取平均值）-Using Matlab programming, to solve the problem of single sample training BP network, set a stop error iterations Emin and the maximum number of iterations. In the debugging process, by continuously adjusting the hidden layer nodes, learning rate η, fast convergence and find a set of parameters small errors. Produce a uniform distribution in the interval [4,4] of the test sample, the model has been established input output with the expected output Hermit polynomial calculated by comparing the total error (run 5 times the average) Platform: |
Size: 11264 |
Author:李昭青 |
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Description: 通过计算机上的多媒体卡，按照下列过程进行声音信号的录制、处理和回放，观察处理前后的波形，同时比较声音信号处理前后的效果。
1、 将录制的信号反褶以后播放（反褶以后的信号移到t=0点开始）。
2、 将录制的信号改变取样率（相当于进行了尺度变化）以后播放。
3、 将录制以后的信号延时1s，并将幅度降低一半，接着与原来的信号相加，然后播放。-Multimedia card on your computer, follow the process for recording, processing, and playback sound signal waveform observed before and after treatment, while comparative effectiveness sound signal before and after treatment. 1, the signal recorded after deconvolution player (deconvolution signal moves after t = start 0:00). 2, the signal is changing the sample rate recording (is equivalent to a scale change) later playback. 3, after the recorded signal delay 1s, and reduce the rate by half, and then with the original signal are added, then play it. Platform: |
Size: 1024 |
Author:陈威宇 |
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Description: TEL7022程序，TXT。TEL7022程序TEL7022程序(USB 2.0 Full-Speed compliant supported USB Audio Class 1.0
16/24 bit Resolutions supported
8/16/32/44.1/48/96 KHz sampling rates supported
2-input channels and 2-output channels supported by one I2S pairs with
independent sample rate
Built in IEC60958 professional S/PDIF TX & RX, AES/EBU supported
8 GPIO pins with interrupt
Built-in ROM for USB descriptor (Support up to 16bit/48KHz only)
Two-wire MPU interface
Built in 1.8V LDO for core voltage
3.3V Operation Voltage
0.18 um CMOS process
LQFP-48 package
SONY Green Partner standard compliant) Platform: |
Size: 2048 |
Author:haikang
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