Sine — Indicators and Strategies

Wave Smoother [WS]The Wave Smoother is a unique FIR filter built from the interaction of two trigonometric waves. A cosine carrier wave is modulated by a sine wave at half the carrier's period, creating smooth transitions and controlled undershoot. The Phase parameter (0° to 119°) adjusts the modulating wave's phase, affecting both response time and undershoot characteristics. At 30° phase the impulse response starts at 0.5 and exhibits gentle undershoot, providing balanced smoothing. Higher phase values reduce ramp-up time and increase undershoot - this undershoot in the impulse response creates overshooting behavior in the filter's output, which helps reduce lag and speed up the response. The default 70° phase setting provides maximum speed while maintaining stability, though practical settings can range from 30° to 70°. The filter's impulse response consists entirely of smooth curves, ensuring consistent behavior across all settings. This design offers traders flexible control over the smoothing-speed trade-off while maintaining reliable signal generation.

Sinusoidal High Pass Filter (SHPF)Sinusoidal High Pass Filter This script implements a sinusoidal high pass filter, which is a type of digital filter that is used to remove low frequency components from a signal. The filter is defined by a series of weights that are applied to the input data, with the weights being determined by a sinusoidal function. The resulting filtered signal is then plotted on a chart, allowing the user to visualize the effect of the filter on the original signal. The script begins by defining the sinusoidal_hpf function, which takes three arguments: _series, _period, and _phase_shift. The _series argument is the input data series that will be filtered, and the _period argument determines the length of the filter. The _phase_shift argument is an optional parameter that allows the user to adjust the phase of the sinusoidal function that is used to calculate the filter weights. The function then initializes a variable ma to 0.0, and loops through each data point in the input series, starting from the most recent and going back in time for the specified _period number of points. For each data point, the function calculates a weight using a sinusoidal function, and adds the weighted data point to the ma variable. Finally, the function returns the average of the weighted data points by dividing ma by the _period. The script also includes user input fields for the Length and Phase Shift parameters, which allows the user to customize the filter according to their specific needs. The filtered signal is then plotted on a chart, along with a reference line at 0. Overall, this script provides a useful tool for analyzing and processing financial data, and can be easily customized to fit the needs of the user.

Sine Wave Theory There are some ideas out there that the market is like a collection of quantum events and that it could all be broken down into sine waves. I created this script to put that to the test. The idea is simple, I tested 3 different factors that could be put into sine wave form. 1.) Bar Change 2.) Volume Average Change 3.) Coin Flip For the bar change, I simply allow the sine wave to move upwards or downwards if the bars have changed color in their sequence. For example, if there were 3 red bars and 1 green bar, it would not move the sine wave up or down until the green bar appeared. For the average volume change, it was the same idea, except that the sine wave could only move up or down if the volume had moved up or below the average value of the length given for calculating the average volume. Finally, the coin flip simply simulates flipping a coin, and allows the sine wave to move one direction or the other once it has a side that is different from the previous chosen side. For example, heads, heads, heads, tails (once it flipped to tails, this would allow it to move a direction). The sine wave trading theory that I watched claimed that if you know the correct sine wave # (which is how large the peak is, and/or the sine wave count which is how many peaks and valleys occur) that you can successfully predict future trades. Their claims that the reason it does not look like a perfect sine wave for these events is because there is different amounts of trading going on, thus the timing will be slightly off. I am posting this to disagree with their ideas. For example, if you select to turn on trading for coin flip and turn off bar change, you will see the coin flip did better on the default settings! It just so happens that any setting will eventually be good, making all the sine wave variations just completely random if you win or not. I posted this to demonstrate how silly trading sine waves is. The real trick is using cosine and tangent waves... lol j/k I hope this helps someone avoid this scam concept.

Clutter-Filtered, D-Lag Reducer, Spec. Ops FIR Filter [Loxx]Clutter-Filtered, D-Lag Reducer, Spec. Ops FIR Filter is a FIR filter moving average with extreme lag reduction and noise elimination technology. This is a special instance of a static weight FIR filter designed specifically for Forex trading. This is not only a useful indictor, but also a demonstration of how one would create their own moving average using FIR filtering weights. This moving average has static period and weighting inputs. You can change the lag reduction and the clutter filtering but you can't change the weights or the numbers of bars the weights are applied to in history. Plot of weighting coefficients used in this indicator These coefficients were derived from a smoothed cardinal sine weighed SMA on EURUSD in Matlab. You can see the coefficients in the code. What is Normalized Cardinal Sine? The sinc function sinc (x), also called the "sampling function," is a function that arises frequently in signal processing and the theory of Fourier transforms. In mathematics, the historical unnormalized sinc function is defined for x ≠ 0 by sinc x = sinx / x In digital signal processing and information theory, the normalized sinc function is commonly defined for x ≠ 0 by sinc x = sin(pi * x) / (pi * x) What is a Generic or Direct Form FIR Filter? In signal processing, a finite impulse response (FIR) filter is a filter whose impulse response (or response to any finite length input) is of finite duration, because it settles to zero in finite time. This is in contrast to infinite impulse response (IIR) filters, which may have internal feedback and may continue to respond indefinitely (usually decaying). The impulse response (that is, the output in response to a Kronecker delta input) of an Nth-order discrete-time FIR filter lasts exactly {\displaystyle N+1}N+1 samples (from first nonzero element through last nonzero element) before it then settles to zero. FIR filters can be discrete-time or continuous-time, and digital or analog. A FIR filter is (similar to, or) just a weighted moving average filter, where (unlike a typical equally weighted moving average filter) the weights of each delay tap are not constrained to be identical or even of the same sign. By changing various values in the array of weights (the impulse response, or time shifted and sampled version of the same), the frequency response of a FIR filter can be completely changed. An FIR filter simply CONVOLVES the input time series (price data) with its IMPULSE RESPONSE. The impulse response is just a set of weights (or "coefficients") that multiply each data point. Then you just add up all the products and divide by the sum of the weights and that is it; e.g., for a 10-bar SMA you just add up 10 bars of price data (each multiplied by 1) and divide by 10. For a weighted-MA you add up the product of the price data with triangular-number weights and divide by the total weight. Ultra Low Lag Moving Average's weights are designed to have MAXIMUM possible smoothing and MINIMUM possible lag compatible with as-flat-as-possible phase response. What is a Clutter Filter? For our purposes here, this is a filter that compares the slope of the trading filter output to a threshold to determine whether to shift trends. If the slope is up but the slope doesn't exceed the threshold, then the color is gray and this indicates a chop zone. If the slope is down but the slope doesn't exceed the threshold, then the color is gray and this indicates a chop zone. Alternatively if either up or down slope exceeds the threshold then the trend turns green for up and red for down. Fro demonstration purposes, an EMA is used as the moving average. This acts to reduce the noise in the signal. What is a Dual Element Lag Reducer? Modifies an array of coefficients to reduce lag by the Lag Reduction Factor uses a generic version of a Kalman velocity component to accomplish this lag reduction is achieved by applying the following to the array: 2 * coeff - coeff The response time vs noise battle still holds true, high lag reduction means more noise is present in your data! Please note that the beginning coefficients which the modifying matrix cannot be applied to (coef whose indecies are < LagReductionFactor) are simply multiplied by two for additional smoothing . Things to note Due to the computational demands of this indicator, there is a bars back input modifier that controls how many bars back the indicator is calculated on. Because of this, the first few bars of the indicator will sometimes appear crazy, just ignore this as it doesn't effect the calculation. Related Indicators STD-Filtered, Ultra Low Lag Moving Average Included Bar coloring Loxx's Expanded Source Types Signals Alerts

STD-Filtered, Ultra Low Lag Moving Average [Loxx]STD-Filtered, Ultra Low Lag Moving Average is a FIR filter that smooths price using a low-pass filtering with weights derived from a normalized cardinal since function. This indicator attempts to reduce lag to an extreme degree. Try this on various time frames with various Type inputs, 0 is the default, so see where the sweet spot is for your trading style. What is a Finite Impulse Response Filter? In signal processing, a finite impulse response (FIR) filter is a filter whose impulse response (or response to any finite length input) is of finite duration, because it settles to zero in finite time. This is in contrast to infinite impulse response (IIR) filters, which may have internal feedback and may continue to respond indefinitely (usually decaying). The impulse response (that is, the output in response to a Kronecker delta input) of an Nth-order discrete-time FIR filter lasts exactly {\displaystyle N+1}N+1 samples (from first nonzero element through last nonzero element) before it then settles to zero. FIR filters can be discrete-time or continuous-time, and digital or analog. A FIR filter is (similar to, or) just a weighted moving average filter, where (unlike a typical equally weighted moving average filter) the weights of each delay tap are not constrained to be identical or even of the same sign. By changing various values in the array of weights (the impulse response, or time shifted and sampled version of the same), the frequency response of a FIR filter can be completely changed. An FIR filter simply CONVOLVES the input time series (price data) with its IMPULSE RESPONSE. The impulse response is just a set of weights (or "coefficients") that multiply each data point. Then you just add up all the products and divide by the sum of the weights and that is it; e.g., for a 10-bar SMA you just add up 10 bars of price data (each multiplied by 1) and divide by 10. For a weighted-MA you add up the product of the price data with triangular-number weights and divide by the total weight. Ultra Low Lag Moving Average's weights are designed to have MAXIMUM possible smoothing and MINIMUM possible lag compatible with as-flat-as-possible phase response. What is Normalized Cardinal Sine? The sinc function sinc(x), also called the "sampling function," is a function that arises frequently in signal processing and the theory of Fourier transforms. In mathematics, the historical unnormalized sinc function is defined for x ≠ 0 by sinc x = sinx / x In digital signal processing and information theory, the normalized sinc function is commonly defined for x ≠ 0 by sinc x = sin(pi * x) / (pi * x) How this works, (easy mode) 1. Use a HA or HAB source type 2. The lower the Type value the smoother the moving average 3. Standard deviation stepping is added to further reduce noise Included Bar coloring Signals Alerts Loxx's Expanded Source Types

Normalized, Variety, Fast Fourier Transform Explorer [Loxx]Normalized, Variety, Fast Fourier Transform Explorer demonstrates Real, Cosine, and Sine Fast Fourier Transform algorithms. This indicator can be used as a rule of thumb but shouldn't be used in trading. What is the Discrete Fourier Transform? In mathematics, the discrete Fourier transform (DFT) converts a finite sequence of equally-spaced samples of a function into a same-length sequence of equally-spaced samples of the discrete-time Fourier transform (DTFT), which is a complex-valued function of frequency. The interval at which the DTFT is sampled is the reciprocal of the duration of the input sequence. An inverse DFT is a Fourier series, using the DTFT samples as coefficients of complex sinusoids at the corresponding DTFT frequencies. It has the same sample-values as the original input sequence. The DFT is therefore said to be a frequency domain representation of the original input sequence. If the original sequence spans all the non-zero values of a function, its DTFT is continuous (and periodic), and the DFT provides discrete samples of one cycle. If the original sequence is one cycle of a periodic function, the DFT provides all the non-zero values of one DTFT cycle. What is the Complex Fast Fourier Transform? The complex Fast Fourier Transform algorithm transforms N real or complex numbers into another N complex numbers. The complex FFT transforms a real or complex signal x in the time domain into a complex two-sided spectrum X in the frequency domain. You must remember that zero frequency corresponds to n = 0, positive frequencies 0 < f < f_c correspond to values 1 ≤ n ≤ N/2 −1, while negative frequencies −fc < f < 0 correspond to N/2 +1 ≤ n ≤ N −1. The value n = N/2 corresponds to both f = f_c and f = −f_c. f_c is the critical or Nyquist frequency with f_c = 1/(2*T) or half the sampling frequency. The first harmonic X corresponds to the frequency 1/(N*T). The complex FFT requires the list of values (resolution, or N) to be a power 2. If the input size if not a power of 2, then the input data will be padded with zeros to fit the size of the closest power of 2 upward. What is Real-Fast Fourier Transform? Has conditions similar to the complex Fast Fourier Transform value, except that the input data must be purely real. If the time series data has the basic type complex64, only the real parts of the complex numbers are used for the calculation. The imaginary parts are silently discarded. What is the Real-Fast Fourier Transform? In many applications, the input data for the DFT are purely real, in which case the outputs satisfy the symmetry X(N-k)=X(k) and efficient FFT algorithms have been designed for this situation (see e.g. Sorensen, 1987). One approach consists of taking an ordinary algorithm (e.g. Cooley–Tukey) and removing the redundant parts of the computation, saving roughly a factor of two in time and memory. Alternatively, it is possible to express an even-length real-input DFT as a complex DFT of half the length (whose real and imaginary parts are the even/odd elements of the original real data), followed by O(N) post-processing operations. It was once believed that real-input DFTs could be more efficiently computed by means of the discrete Hartley transform (DHT), but it was subsequently argued that a specialized real-input DFT algorithm (FFT) can typically be found that requires fewer operations than the corresponding DHT algorithm (FHT) for the same number of inputs. Bruun's algorithm (above) is another method that was initially proposed to take advantage of real inputs, but it has not proved popular. There are further FFT specializations for the cases of real data that have even/odd symmetry, in which case one can gain another factor of roughly two in time and memory and the DFT becomes the discrete cosine/sine transform(s) (DCT/DST). Instead of directly modifying an FFT algorithm for these cases, DCTs/DSTs can also be computed via FFTs of real data combined with O(N) pre- and post-processing. What is the Discrete Cosine Transform? A discrete cosine transform ( DCT ) expresses a finite sequence of data points in terms of a sum of cosine functions oscillating at different frequencies. The DCT , first proposed by Nasir Ahmed in 1972, is a widely used transformation technique in signal processing and data compression. It is used in most digital media, including digital images (such as JPEG and HEIF, where small high-frequency components can be discarded), digital video (such as MPEG and H.26x), digital audio (such as Dolby Digital, MP3 and AAC ), digital television (such as SDTV, HDTV and VOD ), digital radio (such as AAC+ and DAB+), and speech coding (such as AAC-LD, Siren and Opus). DCTs are also important to numerous other applications in science and engineering, such as digital signal processing, telecommunication devices, reducing network bandwidth usage, and spectral methods for the numerical solution of partial differential equations. The use of cosine rather than sine functions is critical for compression, since it turns out (as described below) that fewer cosine functions are needed to approximate a typical signal, whereas for differential equations the cosines express a particular choice of boundary conditions. In particular, a DCT is a Fourier-related transform similar to the discrete Fourier transform (DFT), but using only real numbers. The DCTs are generally related to Fourier Series coefficients of a periodically and symmetrically extended sequence whereas DFTs are related to Fourier Series coefficients of only periodically extended sequences. DCTs are equivalent to DFTs of roughly twice the length, operating on real data with even symmetry (since the Fourier transform of a real and even function is real and even), whereas in some variants the input and/or output data are shifted by half a sample. There are eight standard DCT variants, of which four are common. The most common variant of discrete cosine transform is the type-II DCT , which is often called simply "the DCT". This was the original DCT as first proposed by Ahmed. Its inverse, the type-III DCT , is correspondingly often called simply "the inverse DCT" or "the IDCT". Two related transforms are the discrete sine transform ( DST ), which is equivalent to a DFT of real and odd functions, and the modified discrete cosine transform (MDCT), which is based on a DCT of overlapping data. Multidimensional DCTs ( MD DCTs) are developed to extend the concept of DCT to MD signals. There are several algorithms to compute MD DCT . A variety of fast algorithms have been developed to reduce the computational complexity of implementing DCT . One of these is the integer DCT (IntDCT), an integer approximation of the standard DCT ,: ix, xiii, 1, 141–304 used in several ISO /IEC and ITU-T international standards. What is the Discrete Sine Transform? In mathematics, the discrete sine transform (DST) is a Fourier-related transform similar to the discrete Fourier transform (DFT), but using a purely real matrix. It is equivalent to the imaginary parts of a DFT of roughly twice the length, operating on real data with odd symmetry (since the Fourier transform of a real and odd function is imaginary and odd), where in some variants the input and/or output data are shifted by half a sample. A family of transforms composed of sine and sine hyperbolic functions exists. These transforms are made based on the natural vibration of thin square plates with different boundary conditions. The DST is related to the discrete cosine transform (DCT), which is equivalent to a DFT of real and even functions. See the DCT article for a general discussion of how the boundary conditions relate the various DCT and DST types. Generally, the DST is derived from the DCT by replacing the Neumann condition at x=0 with a Dirichlet condition. Both the DCT and the DST were described by Nasir Ahmed T. Natarajan and K.R. Rao in 1974. The type-I DST (DST-I) was later described by Anil K. Jain in 1976, and the type-II DST (DST-II) was then described by H.B. Kekra and J.K. Solanka in 1978. Notable settings windowper = period for calculation, restricted to powers of 2: "16", "32", "64", "128", "256", "512", "1024", "2048", this reason for this is FFT is an algorithm that computes DFT (Discrete Fourier Transform) in a fast way, generally in 𝑂(𝑁⋅log2(𝑁)) instead of 𝑂(𝑁2). To achieve this the input matrix has to be a power of 2 but many FFT algorithm can handle any size of input since the matrix can be zero-padded. For our purposes here, we stick to powers of 2 to keep this fast and neat. read more about this here: Cooley–Tukey FFT algorithm SS = smoothing count, this smoothing happens after the first FCT regular pass. this zeros out frequencies from the previously calculated values above SS count. the lower this number, the smoother the output, it works opposite from other smoothing periods Fmin1 = zeroes out frequencies not passing this test for min value Fmax1 = zeroes out frequencies not passing this test for max value barsback = moves the window backward Inverse = whether or not you wish to invert the FFT after first pass calculation Related indicators Real-Fast Fourier Transform of Price Oscillator STD-Stepped Fast Cosine Transform Moving Average Real-Fast Fourier Transform of Price w/ Linear Regression Variety RSI of Fast Discrete Cosine Transform Additional reading A Fast Computational Algorithm for the Discrete Cosine Transform by Chen et al. Practical Fast 1-D DCT Algorithms With 11 Multiplications by Loeffler et al. Cooley–Tukey FFT algorithm Ahmed, Nasir (January 1991). "How I Came Up With the Discrete Cosine Transform". Digital Signal Processing. 1 (1): 4–5. doi:10.1016/1051-2004(91)90086-Z. DCT-History - How I Came Up With The Discrete Cosine Transform Comparative Analysis for Discrete Sine Transform as a suitable method for noise estimation

Periodic Channel This indicator try to create a channel by summing a re-scaled and readapted sinusoidal wave form to the price mean. The length parameter control the speed of the sinusoidal wave form, this parameter is not converted to a sine wave period for allowing a better estimation, higher length's work better but feel free to try shorter periods. The invert parameter invert the sinusoidal wave. Each bands represent possible return points, the higher the band the higher the probability. Inverted sin wave exemple The performance of the indicator is subjective to the main estimation (blue line), select the parameter that best fit the blue line to the price. Best ragards

Sine Weighted Moving Average Sine Weighted Moving Average indicator. The result is very similar to the Triangular Moving Average.