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I am trying to read OFDM synchronization paper by Schmidl and Cox. The math is slightly overwhelming. What is the formula that they use in implementation that distinguishes between coarse estimation and fine estimation? Secondly once fine frequency offset is done, how do you determine that you need to do coarse frequ...
I'm looking for an algorithm which can accurately estimate the time-offset/delay between two audio signals in real-time. One audio signal is a loudspeaker signal and the other signal is the microphone signal. Audio is processed in blocks of 10ms (sample frequency 16kHz). There is light office noise embedded in the si...
I want to convert the two's complement, 16-bit output from an ADC into a generic fixed-point format, $\rm Qm.n$, where $\rm m$ is the number of integer bits and $\rm n$ is the number of fraction bits. Any hints/procedure to do this job would be appreciated.
I use the below matlab function to design the unquantised coefficient set: [b,a]=ellip (4,2,60,2*[0.2,0.3]); and then i use the matlab finction "tf2sos" to convert the direct form coefficients to a set of a factorized second-order sections: [sos,g]=tf2sos(b,a); My result is: The normalized version of the differenc...
Windowing is a popular method of reducing the spectral sidelobes of OFDM. A popular window usually used for this purpose in the raised cosine (RC) window, because of its tapered and smooth edges. I understand the concept of windowing and why it reduces the sidelobes, however what I don't understand is this: "windowing ...
Anybody knows what is the best pdf for periodic signal ? I have signal with $480$ samples $\left(16000*0.03=480\right)$ but I could not find the exact pdf for that. sometimes it appear like 2 mixed Gaussian.
I am required extremely fast to fill the gap and learn Fourier Transform with application in Computer Vision. It was easy to find mathematical aspects of Fourier Transform, but I am more interested in application in Computer Vision for a newbie with a lot of examples. If you familiar with a textbook or have any other...
so there is this paper I'm reading and trying to understand fully: Towards Practical Identification of HF RFID Devices http://dl.acm.org/citation.cfm?id=2240276.2240278 I don't want to link the PDF here, but maybe you can catch it from ACM or somewhere else. Now what we need to know: They record a signal for $5 \mu s$ ...
Currently I'm working on a project to record ELF (Extremely Low Frequency) signals from a receiver. The highest frequency of interest is about 30 Hz and the system should have a flat frequency response. I'm looking for a simple solution using an 8 bit ADC and I wonder how the resolution can be improved using oversampl...
I use matlab function tf2sos to convert digital filter transfer function data to second-order sections form: [sos,g] = tf2sos(b,a) Taken from the help menu of the matlab: G is a scalar which accounts for the overall gain of the system My question is: Because G is the overall gain if i draw the block diagram is goin...
I am a long time software engineer but have practically zero experience with signal/audio processing. I am interested in learning about signal processing via a use-case we have for one of our audio components. NOTE: this is just a learning exercise for me... it is not a priority that the end result be useful. We have a...
Sometimes the terms 'Fourier domain', 'complex frequency domain', 'Frequency domain' and 's domain' are used interchangeably. Take those answers here for example: http://answers.yahoo.com/question/index?qid=20090924230321AAxvnJg Can you really use them interchangeably all the time, without being technically wrong? So, ...
I know overlap save and overlap add are used for long data sequence filtering. Are there any other similar or better techniques like these?
I am computing the fourier series of the following function between $[-0.5, 0.5]$ $$\displaystyle f(t) = \frac{1}{2} - |t|$$ According to the definition of Fourier Series the coefficients are given by $$\displaystyle \hat f_n = \frac{-(-1)^n}{2\pi ^2 n^2} + \frac{1}{2\pi ^2 n^2}$$ That's easy to calculate. So far so go...
I am performing a system identification by striking an experimental model with an instrumented impact hammer and measuring the strain response. The timebase for the recorded signals is 5 microseconds and the hammer pulse width is typically around 200 microseconds. The data are recorded for 20 milliseconds prior to the ...
I am confusing when I find the lpc roots magnitude, f1=0.9313, f2=0.9308, f3=0.91 and f4=0.95 but when I look at LPC envelope the f1 is higher than f4 in term of magnitude in dB. Could anybody explain it ? I have wrote following MATLAB code : frame=wave((i-1)*frame_length+1:(i*frame_length)); a=lpc(frame,30); r=roots(a...
I want to implement Wavelet Transform from the scratch, that mean breaking the wavelet transform into its equations to implement in any Programming language. Matlab Comes with built-in functions to implement Wavelet Transform but It is really hard to understand which processes are exactly involved in the implementation...
I asked myself how to compute dBFS (dB full scale) from a value of sample between 1 and -1? and in general?
I use discrete Fourier transform for digital image processing purposes , but I don't understand basic concept behind it. For example : What information exists in frequency domain? What is difference between spatial domain and frequency domain?
The question I have is related to image quality evaluation. Suppose you are supposed to perform OCR (Optical Character Recognition) on a set of document images. Some images, however, are very blurring, and for these images blind deconvolution is needed. However, there are good images that do not need deconvolution. The...
My question is regarding the input size of a signal which is not a power of 2 and we have to take the fft of it. Some solutions say that suppose if we want to take the fft of 1800 we should zero pad it till the length of 2048 to make it power of 2 and then apply the radix 2 algorithm. But there are other solutions as ...
I'm trying to implement a multipath ricean channel model. I had found several resources but I'm kind of lost with too much information. What I have from now is the following step: 1- Generate the signal without noise/channel 2- convolve with the channel response 3- add white gaussian noise My problem is how to genera...
Given a sound clip (whose source is an FM radio signal) I'd like to measure how different from background noise (the "in-between channels" noise) the sound clip is, so that I can compare the "non-noisiness" (and thus the quality) of two radio signals. Is there a good way to do this?
I want to up-sample my input signal by a factor of $2$. I saw zero padding followed by low pass filter method being used in few cases. But still I need some help in this. Say I have $10$ input samples and my interpolating factor is $2$, then if I am doing interpolation using above method, will I get $20$ samples at t...
I am trying to extract only urban aeas from satellite imagery Using Matlab. I have applied some edge detectors like canny and also used morphological functions and I am able to extract all the boundaries ,but not able to seperate them from various urban aeas like roads, trees and waterbodies. Please help i can share th...
I have read in my textbook and i don't understand this part, that the Direct Form II implementation of an IIR filter is a bit tricky in memory addressing.There is need of clever memory addressing. Memory is the place that we store partial summations(states of the filter). Anyone that understands the problem and explai...
Can anyone tell me how does bit reversal order technique works for FFT when you are trying to break down the FFT to small sizes. Like I want to only for the when the index is odd because that involve some reverse carry propagation, but when the index is even you just add N/2 to it to get your reverse index r. An exampl...
I'm trying to understand this code, however I'm unable to deduce the meaning of reds and trans. I would be glad if you can help with it: import sys import Image # pixel constants in hex trans="\x00\x00\x00\x00" # reds most to least, the last is trans and used as a noise factor reds=("\xf1\x34\x00\xff" , "\xfa\x00\x2f\...
I have a wav file which is gained by playing 2 different wav files with 2 speakers and recorded with mic. If I want to get one of the original signals using the resulting wav file and the other original wav file, is it possible to do this? I checked online and only got blind signal separation algorithm. Thank you in ad...
Three methods of estimating maximum signal level at a node: Worst Case Min-Max (L1-Norm) Mean Square (L2-Norm) Frequency Domain (Linfinity-Norm) In matlab i found the code for L1, L2 Hfull=[h(r,:),conj(fliplr(h(r,2:end-1)))]; L1(r)=sum(abs(ifft(Hfull))); L2(r)=sqrt(mean(Hfull.*conj(Hfull))); For Linf is corr...
Question Details: There are a variety of image processing algorithms useful for detecting reference marks (contour tracing, FAST, MSER, Haar Feature-based Cascade Classifier, or even just Harris Edge detector, etc). There are a variety of simple geometric shapes one might pick for use as reference marks (circle, square...
This is just an idea. How can we model the kalman filter to get the state representation in continuous space when the observations to the system are actually from the discrete space. The discrete observation, for example, can be sequence of digits (0-9) where the digits themselves do not have numerical meaning but the ...
Here's my problem. The input signals $x$ and $y$ will be having the time value aligned with each other. However, the data are not evenly sampled. I would like to calculate CPSD of both signals. The solution comes to my mind is as follow $R_{xy}$ = cross correlation of $x$ and $y$ (I'm not sure how to do it with uneven...
I am currently working on the FFT function in MATLAB. In this function, i require a 1024 number of data Time Domain plot and produce a 400 number of data Frequency domain plot. After applying the FFT function, i obtain a result of a graph going up to 6000Hz. However, the Frequency band produced is too big and I only wa...
I'm trying to understand how DFT works exactly. However, when experimenting around, I compared both Matlab generated FFT result with a dummy approach result and I get similar result. However, the Imaginary part of both results are negated. The code below is my implementation. The implementation of dummy approach is bas...
I have a raw infrared image with 16384(2^14) intensity levels.The raw image must be compressed into one with 256(2^8) intensity levels. In order to retain the original details( small blocks) , the compressed image must be enhanced. Which the best method I should select?Someone advice that "histogram matching" method sh...
I want to know how I can determine whether a series of data is periodic or not. I want to use Fourier transform/series. My data looks either aperiodic [111100001111000110010101010000101] or periodic [11001100110011001100] and I need to decide which it is automatically. What types of analyses or calculations can ...
I am trying to implement speech recognition using backpropagation algorithm, and I have been following this paper. I have followed it all the way, except that it tells me to zero-pad the coefficients when there is an empty slot, because the MFCC coefficients of two samples are not the same. Now my question is, do I ra...
I use this snippet of python code to transform data to Fourier phase and magnitude and then retrieving original data. from random import randint as RI import numpy.fft as FFT import math w = 4 h = 4 random_range = 255 vals = [[] for i in range(h)] for i in range(h): for j in range(w): vals[i].append(RI(0...
I am implementing phase correlation algorithm to determine shift between two images. It generally works, but I am not sure how to interpret the resulting shift. Pseudocode: ft1 = FT(input1) ft2 = FT(input2) for each pixel p product = ft1[p] * conj(ft2[p]) cross_power[p] = product / abs(product) cross_correlation ...
I've used Kalman filters for various things in the past, but I'm now interested in using one to track position, speed and acceleration in the context of tracking position for smartphone apps. It strikes me that this should be a text book example of a simple linear Kalman filter, but I can't seem to find any online lin...
I have quite a straight-forward question. What I aim for is the generation of a certain set of random numbers with a normal distribution (mu = 0, sigma = 1). Now, the best way to approach the above mentioned gaussian bell is to generate quite a large number of samples. Unfortunately, in my case, I can only generate a r...
In my filter specs, the passband gain is given as 6.02db. How do I achieve this? Usually all texts give the passband gain as 0db (as well as passband ripples if any).
I want to port the following FFT code on a microcontoller having a very little floating point capability. I want to perform FFT of the output from 12 bit ADC. Since my values are all 12 bit numbers it is wasteful of storage and processing power (I do not need very accurate FFT output) if I do every thing in double as t...
I am an image processing engineer. I my work I often find that there are many aspects for my code to be improved. Who can recommend a book which focuses on optimization methods for image processing? Any advice will be appreciated!
I am confused about this Q&A : Interpolation by factor of 2 If my input signal $$ x[n]= x0,x1,x2,x3 $$ then according to the threads explaination my $v[n]$ will be $$ v[n]=x0,0,x1,0,x2,0,x3,0 $$ $x[n] \rightarrow$ zeropad $\rightarrow v[n]$ $v[n]\rightarrow$ LPF $\rightarrow y[m]$ My $y[m]$ is the output of LPF whic...
I'm trying to work out what the correct solution is for a DCT of a 4x4 checkerboard matrix (let's call it A) and for a matrix of ones (let's call it B). So A is: 0 1 0 1 1 0 1 0 0 1 0 1 1 0 1 0 And B is: 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 1 With Octav...
I am trying to thin foreground pixels of a finger print until they are one pixel wide without using the built in bwmorph and thin operations. Can anyone explain to me how to write them in matlab without actually using the built in function. Thanks.
I have a bunch of 8x8 images which have been vectorized and formed into a 64xN matrix, X. I'd like to take the 2D-FFT of each of these images without reshaping each column of X into an 8x8 image and using fft2. Any suggestions? I've tried playing around with properties of kronecker products and vec() operations but h...
Thanks to everyone who posted comments/answers to my query yesterday (Implementing a Kalman filter for position, velocity, acceleration ). I've been looking at what was recommended, and in particular at both (a) the wikipedia example on one dimensional position and velocity and also another website that considers a si...
Could anybody tell me a method to calculate the "4 Hz modulation energy" for a speech signal? Thanks. EDIT I get some details : 4 Hz modulation energy: Speech has a characteristic energy modulation peak around the 4 Hz syllabic rate [3]. We use a portion of the MFCC algorithm [4] to convert the audio signal into 40 per...
I am working in the field of digital image restoration. I have images that are degraded by camera shake. My images are similar to this blurry photo from camera shake. Can anyone please suggest a better non-blind deconvolution algorithm? I have already checked some of the blind deconvolution algorithms such as Dilip Kri...
The steps of computing the Mel-Frequency Cepstrum Coefficients (MFCC) are: Frame blocking -> Windowing-> abs(DFT) -> Mel filter bank-> Sum coefficients for each filter-> Logarithm -> DCT But what is the purpose of the logarithm step?
I have some data sets on which I applied Gaussian smoothing using [1 4 6 4 1] kernel. In my program I iterated this kernel 50 times on the data sets. But only a few of them could retain their shape, most of them just flattened and lost all the patterns I want to know is there any way I can estimate the number of iterat...
How to calculate auto-correlation of a bpsk modulated signal, or how to calculate expectation value of complex exponential function manually not by using matlab or any other software? For example, if $$x(t)= a\exp(i2\pi ft)+n(t)$$ how would you solve $$\mathrm{E}[x(t)]=?$$ and $$R(t,t_1)= \mathrm{E}[x(t+t_1/2)x'(t-t_...
I have successfully tested 1D phase correlation algorithm to determine vertical shift between two synthetic images. When I moved to real images, however, it is not able to detect translation at all (the peak is located at 0, which is wrong result). I have the following images: And resulting phase correlation (Magnitu...
I have a noisy signal with frequency between 1/3 and 1/5 of the sampling window. DFT has a very low resolution for low frequency components, what is the best way to find the frequence?
When I recreate a signal it seems to look great. But when I try and increase it's frequency (which is done in a for loop and the variable new_freq=2) it starts to drift. See Image below (The top image is 1hz and the bottom image is 2hz. I circled the image with the problem). I also noticed when I increase the sample f...
I have a signal that it is like a Sinusoidal signal with many local maximum, I want to smooth this signal with almost connecting it's local maximum to each other.
Hey guys, what is the difference between b.) and c.)? Why do we need each of them? Excrept from: http://www.dspguide.com/ch8/5.htm (scroll below)
How did letter d.) become letter e.)? How do we perform this averaging process? Its clear the averaging the signal causes some noise to be reduced.
I am working in digital image restoration field and i have studied this is many research papers that presence of saturated pixels violates the assumption which is initially assumed by many blind deconvolution algorithms, that the image formation process is linear... Can anyone please explain that how this rule is viola...
Given a list of overtones (F1, F2, F3, etc), how do I compute the fundamental frequency? Can I do something like F2/F1=F1/F0? Is it the correct method to use?
I have a wav file recorded from my smartphone's mic, and I want to reconstruct the sampled signal and plot the reconstructed signal. After some research and search, I was able to get the following code: [y, fs, nb] = wavread('new.wav'); x=[1:1:length(y)]; z = y(:,[1]); samp=transpose(z); b = 48000/fs; % Quantizing the...
I read this lecture on using PCA/eigenfaces: http://www.cse.psu.edu/~rcollins/CSE486/lecture32.pdf And I want to use this general procedure to recognize a 3D clay model that I took pictures of from multiple angles. I'm having trouble conceptualizing how it's actually implemented in software. I've "trained" the initial ...
I am sampling a signal at 1kHz, which is essentially a single-frequency sinusoid with a DC offset (and a small amount of noise). I don't know the precise frequency of the sinusoid (it is at most 10Hz), and I want to record the mean value of the signal. Currently I take samples for an arbitrary number of seconds and fin...
Assume a camera has no skew, and has square pixels, that is, its camera-calibration matrix, $K$, looks like: $$ K = \left[\begin{matrix} \alpha&0&u_x\\0&\alpha&u_y\\0&0&1 \end{matrix}\right] $$ Intuitively, it seems that it should be possible to recover the pose of a rectangular planar object, $P$, (for example a flat ...
I would like to ask about smoothing data by using Kalman filter. Due to quantization, I have data that is not smooth. How can I smooth this data by using Kalman Filter. For your information, the data is not constant and have some gradient.
I am confused while writing code to implement Fisher’s F-ratio. Suppose I have 4 speakers, each speaker pronounces 2 words and each word has 3 frames and each frame has 30 sub-bands. $u$ is mean of all speakers. Suppose $x_{ij}$ is one sub-band energy of the $j$th speech frame of speaker $i$ with $j= 1\ldots N$, and $i...
I am trying to find the three cervical vertebrae (C2 , C3 , C4)in x-ray videos (gray images) I am using SURF to identify the place that has these 3 vertebrae (big box for all of them ) SURF gives me good result in this case Now I am trying to identify each one of these vertebrae from this big box, but SURF failed in t...
I was asked this question and couldn't come up with an answer on the spot which didn't involve the frequency domain (basically that the co-efficients of the delay sequence are the impulse response of a FIR filter). Does anyone have any insight which makes this process 'obvious'?
I thought that DSP would be done by using FFT of portions of a signal, modify the samples that result from FFT (since they represent the spectrum of our signal + noise) and remove any unwanted signals and than do an inverse FFT to get a time domain representation of the filtered signal (noise has been removed now). Thi...
I am trying to understand the LidarBoost algorithm as explained in this paper (PDF warning). I don't understand how they take the original depth-images $Y_k$ and transform them into the up-sampled images $D_k$. I get how optical flow is used to align the $Y_k$ into a chosen reference frame, but is the transform betwee...
If I record a sound at a rate of 8000 samples per second I know that the highest frequency that can be reproduced from the sound is 4000 Hz. What I want to know is; If the sound contains frequencies above 4000 Hz when I play the sound back what would I hear and what frequencies would be produced at the points wher...
I am trying to implement OFDM on gnuradio. I have two questions. The sampling frequency which is slightly higher than available bandwidth satisfies Nyqvist criterion for complex baseband signals. So why do we need to add oversampling factor again? Suppose say oversampling factor is 2. How does that help us in recover...
I extracted sift feature to construct a bag-of-words model. I did the following to extract the feature vector: Extracted features from the key frames. After the feature extraction I tried using Principle Component Analysis to reduce the size of my features. Query: What further pre-processing steps should I take befor...
Consider a filter which tracks $x_i$, clipping $|x_i - x_{i-1}| >$ some limit, say 1: in: 0 1 2 10 0 0 10 10 10 0 0 0 0 out: 0 1 2 3 2 1 2 3 4 3 2 1 0 -- |delta| <= 1 (in Python with Numpy: cumsum( clip( diff( x ), - limit, limit )) Can anyone point me to a description of such filters, either in theory o...
I'm new to Matlab/Simulink and would like to convert the continuous system below to discrete for eventual conversion to C code. Matlab has a c2d function, but I don't know how to apply it to a simulink system. The c2d function requires the system to be in transfer function or state-space form, neither of which I know ...
I am working for my class project on making animal repellent , I need to know how I could go about taking a normal audio file and shift its frequency/Pitch very high to make it an ultrasonic sound.
I am learning about the Discrete Cosine Transform (DCT) and I read somewhere that, in general, images have very few high frequency components. Can someone give me an intuitive explanation for why this is so? Is this assumption violated in any scenario?
I have a signal that was recorded with around 300 samples in a second. Now I want to run a low pass and a high pass butterworth filter over it. The frequency for the high pass filter is around 0.05 Hz and the one for the low pass around 5 Hz. But how can I decide which order my butterworth filters should have? Is there...
This is a classical image segmentation problem: I need to make a visual measurement of particles. I've been using watershed, kmeans and mean shift but it doesn't work well on this image. Could you please recommend me some other methods/open source code? P/S: Any manual/semi automatic segmentation is also ok with me. I ...
How can I calculate the (black body equivalent) color temperature of the light source illuminating a given image? Below are screenshots of Adobe Lightroom manipulating color temperature and a shift in the RGB Histograms. Given RGB components of the image, how do I go about calculating it? I should expect a single value...
Why do we need to flip the kernel in 2D convolution in the first place? What's the benefit of this? So, why can't we leave it unflipped? What kind of terrible thing can happen if you don't flip it? SEE: "First, flip the kernel, which is the shaded box, in both horizontal and vertical direction" http://www.songho.ca/dsp...
So I am working on a project using digital signal processing, which I am not familiar with at all, and a lot of this is all new to me. I am using SciPy\NumPy to play around with code to understand the Fourier Transform. Basically, I have been able to read in a wave file and perform an FFT on the data using the followi...
Suppose there is a distribution that has values sampled on the interval 1-25 with corresponding sample values that have to be smoothed. For example: n f[n] 3 123 5 234 8 246 10 567 14 253 18 251 24 132 The first column has sample points between 1-25 but does not have samples for all points on the ...
i am working in digital image restoration field, i am familiar with java and i want to use java in image restoration, can any please suggest me any book or tutorials where i can learn about image restoration using java. I have already checked this java tutorial
I am doing some massive number crunching in MATLAB which involves millions of PSD estimations. Each data segment has length 41. So I have been using the multi-taper method with nfft=41 (details below in the code). My last simulation took more than eight hours. The code is optimized and vectorized as I can make it so I ...
I've sampled a propagating wave so that I know its amplitude at several equidistant points on a line, and at several points in time. That is, I have this discrete function: f[t,x] Now, I want to find the frequency of the wave. But there seems to be two frequencies I can calculate: by finding the FFT at different point...
I know there's a benefit of knowing the Fourier Transform for image processing, but is there a benefit to know Fourier series, or could you just skip them? Would you recommend skipping Fourier series or would that make it too abstract? What's your opinion on this?
I'm modulating a message with particular baudrate, say 2MHz using BPSK , Carrier frequency(Fc) is 8MHz, What should be appropriate Sampling Frequency(Fs) following Nyquist criteria, Should it be greater than 2*(2*baudrate) or 2*(Fc+baudrate)?
Wikipedia lists a number of different FFT algorithms: Cooley–Tukey FFT algorithm, Prime-factor FFT algorithm, Bruun's FFT algorithm, Rader's FFT algorithm, and Bluestein's FFT algorithm What are the pros and cons of each algorithms? (think about: computational expense, parallelizability, ease of implementation, etc.) A...
I need to smooth some data segments and one of the techniques I am experimenting is a low-pass filter. So I designed a low-pass filter in matlab and applied it to my signals. Here is what happened with two of them. Panel A is one of the data segments plotted with its filtered signal. It looks good. This is what I expe...
I have a small 2D Gaussian image filter that I am currently applying as a window by convolution. I would like to do this in the Fourier domain as a single multiplication. I would like to convert my current small filter to the Fourier domain as it is with maximum possible fidelity. I do not know the size of the image th...
I am going through an old exam paper and trying to understand this question where they ask us to tell them what effect the following image filter would have if its convolved with an image: 0 0 0 0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 ...
Can you help me (show me how) to solve the following exercise? For the process with transfer function $G(s)=e^{-2s}/(s+1)$ design a digital controller with sampling time $T_{s} = 1$ that meets the specifications: $e_{∞}$ (steady state error) null whit a step as input; dominant poles in closed loop with damping $ζ≥0,5$...
I am trying to understand how image processing is applied for detection of cancer tumor using image processing. Towards this direction I am trying to get latest information i.e state of the art in this area. I found one book titled "Image-Processing Techniques for Tumor Detection" But this is more than 10 years old an...
I have a degree in Electrical Engineering (focusing on VLSI). However, I am more mathematically minded, and only recently discovered that image processing seems to contain lots of interesting maths. I am looking to break into this field in a about a year or so in spite of not having any formal experience in this area. ...
This link contains the following statement: In contrast to the frequency domain, it is possible to implement non-linear filters in the spatial domain. In this case, the summations in the convolution function are replaced with some kind of non-linear operator. Why is it not possible to perform non-linear filtering in ...
Update 8 Unhappy of having to upload tracks to the service and looking at the new release of RekordBox 3 I've decided to take another look for an offline approach and a finer resolution :D Sounds promising though still in a very alpha state: Johnick - Good Time Note that there's no logarithmic scale nor palette tunin...