c++ (Qt)演示端口，交互式参数

我已经将这个算法的演示应用程序移植到c++ (Qt)上。

代码可以在GitHub上找到这里。带有安装程序的Windows(64位)构建在发布页面上。最后，我将添加一些文档和其他发布版本。

您不能绘制点，但可以从文本文件中导入它们(用空格分隔点——换行也算作空格)。您还可以调整算法参数，实时查看效果。这对于针对特定数据集调整算法以及探索参数如何影响结果非常有用。

上面的截图有些过时;从那以后，我添加了两个原始算法中没有的实验性选项:

反向处理数据集的选项(似乎至少改善了功率谱的结果)。 选项，为峰值设置硬性最小阈值。

我还在窗口中间添加了一个笨拙的缩放/平移条，只需用鼠标拖动它来缩放和平移。

模糊的构建指令:

在发布页面上有一个Windows安装程序(64位)，但如果你想从源代码构建它，要点是:

安装Qt的构建工具，然后将qmake && make放在与.pro文件相同的目录下，或者 安装Qt Creator，打开.pro文件，选择任何默认的构建配置，然后按下构建和/或运行按钮(Creator的左下角)。

我只测试过Qt5。我有91%的信心，如果你手动配置组件，Qt Creator安装程序会让你安装Qt5(如果你手动配置组件，你还需要确认是否安装了Qt Charts)。Qt6可能是一个流畅的构建，也可能不是。有一天，我将测试Qt4和Qt6，使这些文档更好。也许吧。

一种方法是根据以下观察来检测峰:

时间t是一个峰值(y (t) > y (t - 1)) & & ((t) > y (t + 1))

它通过等待上升趋势结束来避免误报。它并不完全是“实时”的，因为它会比峰值差一个dt。灵敏度可以通过要求比较的裕度来控制。在噪声检测和时延检测之间存在一种折衷。 您可以通过添加更多参数来丰富模型:

峰如果y (y (t) - (t-dt) > m) && (y (t) - y (t + dt) > m)

dt和m是控制灵敏度和延时的参数

这是你用上述算法得到的结果:

下面是在python中重现图的代码:

import numpy as np
import matplotlib.pyplot as plt
input = np.array([ 1. ,  1. ,  1. ,  1. ,  1. ,  1. ,  1. ,  1.1,  1. ,  0.8,  0.9,
    1. ,  1.2,  0.9,  1. ,  1. ,  1.1,  1.2,  1. ,  1.5,  1. ,  3. ,
    2. ,  5. ,  3. ,  2. ,  1. ,  1. ,  1. ,  0.9,  1. ,  1. ,  3. ,
    2.6,  4. ,  3. ,  3.2,  2. ,  1. ,  1. ,  1. ,  1. ,  1. ])
signal = (input > np.roll(input,1)) & (input > np.roll(input,-1))
plt.plot(input)
plt.plot(signal.nonzero()[0], input[signal], 'ro')
plt.show()

通过设置m = 0.5，你可以得到一个更清晰的信号，只有一个假阳性:

如果边界值或其他标准取决于未来值，那么唯一的解决方案(没有时间机器，或其他关于未来值的知识)是推迟任何决定，直到有足够的未来值。如果你想要一个高于均值的水平，例如，20点，那么你必须等到你至少有19点才能做出任何峰值决策，否则下一个新点可能会完全超过你19点之前的阈值。

Added: If the statistical distribution of the peak heights could be heavy tailed, instead of Uniform or Gaussian, then you may need to wait until you see several thousand peaks before it starts to become unlikely that a hidden Pareto distribution won't produce a peak many times larger than any you currently have seen before or have in your current plot. Unless you somehow know in advance that the very next point can't be 1e20, it could appear, which after rescaling your plot's Y dimension, would be flat up until that point.

这是一个Python实现的鲁棒峰值检测算法算法。

初始化和计算部分被分开，只有filtered_y数组被保留，它的最大大小等于延迟，因此内存没有增加。(结果与上述答案相同)。 为了绘制图形，还保留了标签数组。

我做了一个github要点。

import numpy as np
import pylab

def init(x, lag, threshold, influence):
    '''
    Smoothed z-score algorithm
    Implementation of algorithm from https://stackoverflow.com/a/22640362/6029703
    '''

    labels = np.zeros(lag)
    filtered_y = np.array(x[0:lag])
    avg_filter = np.zeros(lag)
    std_filter = np.zeros(lag)
    var_filter = np.zeros(lag)

    avg_filter[lag - 1] = np.mean(x[0:lag])
    std_filter[lag - 1] = np.std(x[0:lag])
    var_filter[lag - 1] = np.var(x[0:lag])

    return dict(avg=avg_filter[lag - 1], var=var_filter[lag - 1],
                std=std_filter[lag - 1], filtered_y=filtered_y,
                labels=labels)


def add(result, single_value, lag, threshold, influence):
    previous_avg = result['avg']
    previous_var = result['var']
    previous_std = result['std']
    filtered_y = result['filtered_y']
    labels = result['labels']

    if abs(single_value - previous_avg) > threshold * previous_std:
        if single_value > previous_avg:
            labels = np.append(labels, 1)
        else:
            labels = np.append(labels, -1)

        # calculate the new filtered element using the influence factor
        filtered_y = np.append(filtered_y, influence * single_value
                               + (1 - influence) * filtered_y[-1])
    else:
        labels = np.append(labels, 0)
        filtered_y = np.append(filtered_y, single_value)

    # update avg as sum of the previuos avg + the lag * (the new calculated item - calculated item at position (i - lag))
    current_avg_filter = previous_avg + 1. / lag * (filtered_y[-1]
            - filtered_y[len(filtered_y) - lag - 1])

    # update variance as the previuos element variance + 1 / lag * new recalculated item - the previous avg -
    current_var_filter = previous_var + 1. / lag * ((filtered_y[-1]
            - previous_avg) ** 2 - (filtered_y[len(filtered_y) - 1
            - lag] - previous_avg) ** 2 - (filtered_y[-1]
            - filtered_y[len(filtered_y) - 1 - lag]) ** 2 / lag)  # the recalculated element at pos (lag) - avg of the previuos - new recalculated element - recalculated element at lag pos ....

    # calculate standard deviation for current element as sqrt (current variance)
    current_std_filter = np.sqrt(current_var_filter)

    return dict(avg=current_avg_filter, var=current_var_filter,
                std=current_std_filter, filtered_y=filtered_y[1:],
                labels=labels)

lag = 30
threshold = 5
influence = 0

y = np.array([1,1,1.1,1,0.9,1,1,1.1,1,0.9,1,1.1,1,1,0.9,1,1,1.1,1,1,1,1,1.1,0.9,1,1.1,1,1,0.9,
       1,1.1,1,1,1.1,1,0.8,0.9,1,1.2,0.9,1,1,1.1,1.2,1,1.5,1,3,2,5,3,2,1,1,1,0.9,1,1,3,
       2.6,4,3,3.2,2,1,1,0.8,4,4,2,2.5,1,1,1])

# Run algo with settings from above
result = init(y[:lag], lag=lag, threshold=threshold, influence=influence)

i = open('quartz2', 'r')
for i in y[lag:]:
    result = add(result, i, lag, threshold, influence)

# Plot result
pylab.subplot(211)
pylab.plot(np.arange(1, len(y) + 1), y)
pylab.subplot(212)
pylab.step(np.arange(1, len(y) + 1), result['labels'], color='red',
           lw=2)
pylab.ylim(-1.5, 1.5)
pylab.show()

下面是@Jean-Paul为Arduino微控制器设计的平滑z分数的C语言实现，用于获取加速度计读数，并判断撞击的方向是来自左边还是右边。这表现得非常好，因为这个设备返回一个反弹信号。这是设备对峰值检测算法的输入-显示了来自右边的冲击，然后是来自左边的冲击。你可以看到最初的峰值然后传感器的振荡。

#include <stdio.h>
#include <math.h>
#include <string.h>


#define SAMPLE_LENGTH 1000

float stddev(float data[], int len);
float mean(float data[], int len);
void thresholding(float y[], int signals[], int lag, float threshold, float influence);


void thresholding(float y[], int signals[], int lag, float threshold, float influence) {
    memset(signals, 0, sizeof(int) * SAMPLE_LENGTH);
    float filteredY[SAMPLE_LENGTH];
    memcpy(filteredY, y, sizeof(float) * SAMPLE_LENGTH);
    float avgFilter[SAMPLE_LENGTH];
    float stdFilter[SAMPLE_LENGTH];

    avgFilter[lag - 1] = mean(y, lag);
    stdFilter[lag - 1] = stddev(y, lag);

    for (int i = lag; i < SAMPLE_LENGTH; i++) {
        if (fabsf(y[i] - avgFilter[i-1]) > threshold * stdFilter[i-1]) {
            if (y[i] > avgFilter[i-1]) {
                signals[i] = 1;
            } else {
                signals[i] = -1;
            }
            filteredY[i] = influence * y[i] + (1 - influence) * filteredY[i-1];
        } else {
            signals[i] = 0;
        }
        avgFilter[i] = mean(filteredY + i-lag, lag);
        stdFilter[i] = stddev(filteredY + i-lag, lag);
    }
}

float mean(float data[], int len) {
    float sum = 0.0, mean = 0.0;

    int i;
    for(i=0; i<len; ++i) {
        sum += data[i];
    }

    mean = sum/len;
    return mean;


}

float stddev(float data[], int len) {
    float the_mean = mean(data, len);
    float standardDeviation = 0.0;

    int i;
    for(i=0; i<len; ++i) {
        standardDeviation += pow(data[i] - the_mean, 2);
    }

    return sqrt(standardDeviation/len);
}

int main() {
    printf("Hello, World!\n");
    int lag = 100;
    float threshold = 5;
    float influence = 0;
    float y[]=  {1,1,1.1,1,0.9,1,1,1.1,1,0.9,1,1.1,1,1,0.9,1,1,1.1,1,1,1,1,1.1,0.9,1,1.1,1,1,0.9,
  ....
1,1.1,1,1,1.1,1,0.8,0.9,1,1.2,0.9,1,1,1.1,1.2,1,1.5,1,3,2,5,3,2,1,1,1,0.9,1,1,3,       2.6,4,3,3.2,2,1,1,0.8,4,4,2,2.5,1,1,1,1.2,1,1.5,1,3,2,5,3,2,1,1,1,0.9,1,1,3,
       2.6,4,3,3.2,2,1,1,0.8,4,4,2,2.5,1,1,1}

    int signal[SAMPLE_LENGTH];

    thresholding(y, signal,  lag, threshold, influence);

    return 0;
}

她的结果是影响= 0

不是很好，但这里的影响力= 1

这很好。

下面是这个答案的平滑z-score算法的c++实现

std::vector<int> smoothedZScore(std::vector<float> input)
{   
    //lag 5 for the smoothing functions
    int lag = 5;
    //3.5 standard deviations for signal
    float threshold = 3.5;
    //between 0 and 1, where 1 is normal influence, 0.5 is half
    float influence = .5;

    if (input.size() <= lag + 2)
    {
        std::vector<int> emptyVec;
        return emptyVec;
    }

    //Initialise variables
    std::vector<int> signals(input.size(), 0.0);
    std::vector<float> filteredY(input.size(), 0.0);
    std::vector<float> avgFilter(input.size(), 0.0);
    std::vector<float> stdFilter(input.size(), 0.0);
    std::vector<float> subVecStart(input.begin(), input.begin() + lag);
    avgFilter[lag] = mean(subVecStart);
    stdFilter[lag] = stdDev(subVecStart);

    for (size_t i = lag + 1; i < input.size(); i++)
    {
        if (std::abs(input[i] - avgFilter[i - 1]) > threshold * stdFilter[i - 1])
        {
            if (input[i] > avgFilter[i - 1])
            {
                signals[i] = 1; //# Positive signal
            }
            else
            {
                signals[i] = -1; //# Negative signal
            }
            //Make influence lower
            filteredY[i] = influence* input[i] + (1 - influence) * filteredY[i - 1];
        }
        else
        {
            signals[i] = 0; //# No signal
            filteredY[i] = input[i];
        }
        //Adjust the filters
        std::vector<float> subVec(filteredY.begin() + i - lag, filteredY.begin() + i);
        avgFilter[i] = mean(subVec);
        stdFilter[i] = stdDev(subVec);
    }
    return signals;
}