I want to do a comparison of 2 audio files (each audio file is speaking "ba a ta") with the existing function in matlab called Dynamic Time Warping (DTW). Before doing a dynamic time warping, I get an array/vector from the Fast Fourier Transform (FFT) functions available in matlab, my code so far (my matlab filename: test.m):
fftRecording1 = fft(audioread('C:\Users\handy\Documents\MATLAB\my_recording_1.wav'));
fftRecording2 = fft(audioread('C:\Users\handy\Documents\MATLAB\fajar.wav'));
dist = dtw(fftRecording1, fftRecording2);
When I try the DTW function there is an error because the length (row) of the array/vector 2 file is different. Error message:
Error using dtw (line 82)
The number of rows between X and Y must be equal when X and Y are matrices
Error in test (line 3)
dist = dtw(fftRecording1, fftRecording2);
contents of the fftRecording1 and fftRecording2 variables
My question is: before do the FFT and DTW, how do step by step normalize so that the length (row) 2 audio files is equal? or there are other ways to make the data length (row) 2 audio files is equal?
According to dtw's documentation:
To stretch the inputs, dtw repeats each element of x and y as many times as necessary. If x and y are matrices, then dist stretches them by repeating their columns. In that case, x and y must have the same number of rows.
In your case your columns represent the audio channels, with the rows representing the quantity to be aligned (i.e. the reverse of what dtw is expecting). To setup the inputs according to what dtw expect, simply transpose the inputs:
dist = dtw(transpose(fftRecording1), transpose(fftRecording2));
Dynamic Time Warping does not need the input sequences to be of same length. DTW is actually used to find similarity between two different time aligned sequences.
No, they don’t need to have the same length in a time-related-sense. They need to have the same number of dimensions (2D Signal, 3D Signal,...) which is equivalent to their number or rows. The whole idea of DTW is to match similar contents which might be stretched to different lengths - so there would absolutely be no point in requiring the inputs to have the same length.
Related to your question: just call the dtw with the transposed of your signals and you will get a proper result.
dtw(signal1’, signal2’);
You should apply the DTW on the original signals rather than the fourier transforms. The FFT transfers the signal from time to frequency domain. So instead of warping signal1 in order to match signal2, you are warping frequencies when using FFT before DTW. The amplitude of the fourier transform depends on the number of points in the considered FFT-Time-Window. From my point of view there is absolutely no point in applying DTW on a fourier transform.
Related
This question relates to SciPy's Short-time Fourier Transform function for signal processing.
For some reason I don't understand, the size of the output 'array of sample frequencies' is exactly equal to the hop size. From the documentation:
nperseg : int, optional
Length of each segment. Defaults to 256.
noverlap : int, optional
Number of points to overlap between segments. If None, noverlap = nperseg // 2. Defaults to None. When specified, the COLA constraint must be met (see Notes below).
f : ndarray
Array of sample frequencies.
hop size H = nperseg - noverlap
I'm new to signal processing and Fourier transforms, but as far as I understand a STFT is just chopping an audio file into segments ('time frames') on which you perform a Fourier transform. So if I want to do a STFT on 100 time frames, I'd expect the output to be a matrix of size 100 x F, where F is an array of measured frequencies ('measured' probably isn't the right word here but you know what I mean).
This is kinda what SciPy's implementation does, but the size of f here is what bothers me. It's supposed to be an array describing the different frequencies, like [0Hz 500Hz 1000Hz], and it does, but for some reasons its size exactly the same as the hop size. If the hop size is 700, the number of measured frequencies is 700.
The hop size is the number of samples (i.e. time) between each time frame, and is correctly calculated as H = nperseg - noverlap, but what does this have to do with the frequency array?
Edit: Related to this question
An FFT is an square matrix transform from one orthogonal basis to another of the same dimension. This is because N is the exact number of orthogonal (e.g. that don't interfere with one another) complex sinusoids that fit in a time domain vector of length N.
A longer time vector can contain more frequency information (e.g. it's hard to tell 2 frequencies apart using just 3 sample points, but much easier with 3000 samples, etc.)
You can zero-pad your short time vector of length N to use a longer FFT, but that is identical to interpolating a nice curve between N frequency points, which makes all the FFT results interdependent.
For many purposes (visualization, etc.) an STFT is overlapped, where the adjacent segments share some overlapped data instead of just being end-to-end. This gives better time locality (e.g. the segments can be spaced closer but still be long enough so that each one can provide the frequency resolution required).
I have asked this question on DSP.SE before, but my question has got no attention. Maybe it was not so related to signal processing.
I needed to divide a discrete audio signal into segments to have some statistical processing and analysis on them. Therefore, segments with fixed local mean would be very helpful for my case. Length of segments are predefined, e.g. 512 samples.
I have tried several things. I do use reshape() function to divide audio signal into segments, and then calculate means of every segment as:
L = 512; % Length of segment
N = floor(length(audio(:,1))/L); % Number of segments
seg = reshape(audio(1:N*L,1), L, N); % Reshape into LxN sized matrix
x = mean(seg); % Calculate mean of each column
Subtracting x(k) from each seg(:,k) would make each local mean zero, yet it would distort audio signal a lot when segments are joined back.
So, since mean of hanning window is almost 0.5, substracting 2*x(k)*hann(L) from each seg(:,k) was the first thing I tried. But this time multiplying by 2 (to make the mean of hanning window be almost equal to 1) distorted the neighborhood of midpoints in each segments itself.
Then, I have used convolution by a smaller hanning window instead of multiplying directly, and subtracting these (as shown in figure below) from each seg(:,k).
This last step gives better results, yet it is still not very useful when segments are smaller. I have seen many amazing approaches here on this site for different problems. So I just wonder if there is any clever ways or existing methods to obtain zero local means which distorts an audio signal less. I read that, this property is useful in some decompositions such as EMD. So maybe I need such decompositions?
You can try to use a moving average filter:
x = cumsum(rand(15*512, 1)-0.5); % generate a random input signal
mean_filter = 1/512 * ones(1, 512); % generate a mean filter
mean = filtfilt(mean_filter, 1, x); % filtfilt is used instead of filter to obtain a symmetric moving average.
% plot the result
figure
subplot(2,1,1)
plot(x);
hold on
plot(mean);
subplot(2,1,2)
plot(x - mean);
You can tune the filter by changing the interval of the mean filter. Using a smaller interval, results in lower means inside each interval, but filters also more low frequencies out of your signal.
I am new in signal processing. I want to check the relation between the two wind speed data at different location. i am not getting whether which matlab command I have to use whether it is 'xcorr' or 'cross corr' in matlab ?
While xcorr calculates the Correlation between 2 vectors (By the way, doing it using fft and not conv) crosscorr calculates the Statistics Correlation, namely by removing the means of the samples and standardization:
output = <(x - mean(x)), (y - mean(y))> / (|x| * |y|)
If the vectors which are the input to the functions are centered (Namely with Zero Mean) and normalized they will be equal.
They should be the same but crosscorr also plots the result.
I want to compute a general PCA matrix for a dataset, and I will use it to reduce dimensions of sift descriptors. I have already found some algorithms to compute it, but I couldn't find a way to compute it by using MATLAB.
Can someone help me?
[coeff, score] = princomp(X)
is the right thing to do, but knowing how to use it is a little tricky.
My understanding is that you did something like:
sift_image = sift_fun(img)
which gives you a binary image: sift_feature?
(Even if not binary, this still works.)
Inputs, formulating X:
To use princomp/pca formulate X so that each column is a numel(sift_image) x 1 vector (i.e. sift_image(:))
Do this for all your images and line them up as columns in X. So X will be numel(sift_image) x num_images.
If your images aren't the same size (e.g. pixel dimensions different, more or less of a scene in the images), then you'll need to bring them into some common space, which is a whole different problem.
Unless your stuff is binary, you'll probably want to de-mean/normalize X, both in the column direction (i.e. normalizing each individual image) and row direction (de-meaning the whole dataset).
Outputs
score is the set of eigen vectors: it will be num_pixels * num_images.
To get, say the first eigen vector back into an image shape, do:
first_component = reshape(score(:,1),size(im));
And so on for the rest of the components. There are as many components as input images.
Each row of coeff is the num_images (equal to num_components) set of weights that can be applied to generate each input image. i.e.
input_image_1 = reshape(score * coeff(:,1) , size(original_im));
where input_image_1 is the correct, original shape
coeff(1,:) is a vector (num_images x 1)
score is pixels x num_images
(Disclaimer: I may have the columns/rows mixed up, but the descriptions are correct.)
Does that help?
If you have access to Statistics Toolbox, you can use the command princomp, or in recent versions the command pca.
I currently have a large matrix M (~100x100x50 elements) containing both positive and negative values. At the moment, if I want to smooth this matrix, I use the smooth3 function to apply a gaussian kernel over the entire 3-D matrix.
What I want to achieve is a variable level of smoothing within this matrix - i.e.. different parts of the matrix M are smoothed to different levels of sigma depending of the value in a similar 3-D matrix, d (with values ranging from 0 to 1). Where d is 0, no smoothing occurs, where d is 1 a maximum level of smoothing occurs.
The fact that the matrix is 3-D is trivial. Smoothing in 3 dimensions is nice, but not essential, and my current code (performing various other manipulations) handles each of the 50 slices of M separately anyway. I am happy to replace smooth3 with a convolution of M with a gaussian function, and perform this convolution over each slice individually. What I can't figure out is how to vary the sigma level of this gaussian function (based on d) given its location in M and output the result accordingly.
An alternative approach may be to use matrix d as a mask for a very smooth version of matrix Ms and somehow manipulate M and Ms to give an equivalent result, however I'm not convinced that this will work as I can't think of a function to combine M and Md that won't give artefacts of each of M or Ms when 0 < d < 1...any thoughts?
[I'm using 2009b, and only have access to the Signal Processing toolbox.]
You should have a look at the Guided Image Filter. It is a computationally efficient generalization of the bilateral filter.
http://research.microsoft.com/en-us/um/people/jiansun/papers/guidedfilter_eccv10.pdf
It will allow you to do proper smoothing based on your guidance matrix.