Finding the zeroes of a position vector - python

For my dynamics course I am tasked with writing a python code that will plot the trajectory of a position vector from when it starts on the ground to when it lands on the ground. I currently have my code create a linear space from the two zero values that I calculated by hand, but I want to code that in. Because I also need to create velocity vectors on the trajectory, I have the position vector broken into its x and y components. I have looked into xlim and this thread, but couldn't figure out how to implement them. I'm fairly new to python and coding in general, so I'm still trying to learn how things work.
import numpy as np
import matplotlib.pyplot as plt
#creates a function that returns the x component
def re10(x):
r1 = 0.05*x
return r1
#creates a function that returns the y component
def re20(x):
r2 = -4.91*(x**2) + 30*x + 100
return r2
#Calculates the two zeroes of the trajectory
tmin = (-30 + np.sqrt(30**2 -4*-4.91*100))/(2*-4.91)
tmax = (-30 - np.sqrt(30**2 -4*-4.91*100))/(2*-4.91)
#Initializing time space
t = np.linspace(tmin, tmax, 100)
#Plot
plt.plot(re10(t), re20(t)) #(x, y)

You can easily find the zeroes of a funtion using numpy library.
First, install it. Open a cmd console and write pip install numpy.
Then, write this code in your script:
import numpy
re20 = [-4.91, 30, 100]
zeroes = numpy.roots(coeff)
print(zeroes[0])
print(zeroes[1])
As you will see when running the script from console (or you IDE), numpy.roots(function) will return you the zeroes of your function, as an array.
That is why you use the [] operator to access each one of them (take note that in programming, an array's first element will be at index 0).
To use it directly into your code, you can do:
tmin = zeroes[0]
tmax = zeroes[1]
Simpy is for symbolic operations, it is pretty powerful, but you don't need it for this, my mistake.
Hope you have fun with Python, it's a cool language !

Related

Change the melody of human speech using FFT and polynomial interpolation

I'm trying to do the following:
Extract the melody of me asking a question (word "Hey?" recorded to
wav) so I get a melody pattern that I can apply to any other
recorded/synthesized speech (basically how F0 changes in time).
Use polynomial interpolation (Lagrange?) so I get a function that describes the melody (approximately of course).
Apply the function to another recorded voice sample. (eg. word "Hey." so it's transformed to a question "Hey?", or transform the end of a sentence to sound like a question [eg. "Is it ok." => "Is it ok?"]). Voila, that's it.
What I have done? Where am I?
Firstly, I have dived into the math that stands behind the fft and signal processing (basics). I want to do it programatically so I decided to use python.
I performed the fft on the entire "Hey?" voice sample and got data in frequency domain (please don't mind y-axis units, I haven't normalized them)
So far so good. Then I decided to divide my signal into chunks so I get more clear frequency information - peaks and so on - this is a blind shot, me trying to grasp the idea of manipulating the frequency and analyzing the audio data. It gets me nowhere however, not in a direction I want, at least.
Now, if I took those peaks, got an interpolated function from them, and applied the function on another voice sample (a part of a voice sample, that is also ffted of course) and performed inversed fft I wouldn't get what I wanted, right?
I would only change the magnitude so it wouldn't affect the melody itself (I think so).
Then I used spec and pyin methods from librosa to extract the real F0-in-time - the melody of asking question "Hey?". And as we would expect, we can clearly see an increase in frequency value:
And a non-question statement looks like this - let's say it's moreless constant.
The same applies to a longer speech sample:
Now, I assume that I have blocks to build my algorithm/process but I still don't know how to assemble them beacause there are some blanks in my understanding of what's going on under the hood.
I consider that I need to find a way to map the F0-in-time curve from the spectrogram to the "pure" FFT data, get an interpolated function from it and then apply the function on another voice sample.
Is there any elegant (inelegant would be ok too) way to do this? I need to be pointed in a right direction beceause I can feel I'm close but I'm basically stuck.
The code that works behind the above charts is taken just from the librosa docs and other stackoverflow questions, it's just a draft/POC so please don't comment on style, if you could :)
fft in chunks:
import numpy as np
import matplotlib.pyplot as plt
from scipy.io import wavfile
import os
file = os.path.join("dir", "hej_n_nat.wav")
fs, signal = wavfile.read(file)
CHUNK = 1024
afft = np.abs(np.fft.fft(signal[0:CHUNK]))
freqs = np.linspace(0, fs, CHUNK)[0:int(fs / 2)]
spectrogram_chunk = freqs / np.amax(freqs * 1.0)
# Plot spectral analysis
plt.plot(freqs[0:250], afft[0:250])
plt.show()
spectrogram:
import librosa.display
import numpy as np
import matplotlib.pyplot as plt
import os
file = os.path.join("/path/to/dir", "hej_n_nat.wav")
y, sr = librosa.load(file, sr=44100)
f0, voiced_flag, voiced_probs = librosa.pyin(y, fmin=librosa.note_to_hz('C2'), fmax=librosa.note_to_hz('C7'))
times = librosa.times_like(f0)
D = librosa.amplitude_to_db(np.abs(librosa.stft(y)), ref=np.max)
fig, ax = plt.subplots()
img = librosa.display.specshow(D, x_axis='time', y_axis='log', ax=ax)
ax.set(title='pYIN fundamental frequency estimation')
fig.colorbar(img, ax=ax, format="%+2.f dB")
ax.plot(times, f0, label='f0', color='cyan', linewidth=2)
ax.legend(loc='upper right')
plt.show()
Hints, questions and comments much appreciated.
The problem was that I didn't know how to modify the fundamental frequency (F0). By modifying it I mean modify F0 and its harmonics, as well.
The spectrograms in question show frequencies at certain points in time with power (dB) of certain frequency point.
Since I know which time bin holds which frequency from the melody (green line below) ...
....I need to compute a function that represents that green line so I can apply it to other speech samples.
So I need to use some interpolation method which takes as parameters the sample F0 function points.
One need to remember that degree of the polynomial should equal to the number of points. The example doesn't have that unfortunately, but the effect is somehow ok as for the prototype.
def _get_bin_nr(val, bins):
the_bin_no = np.nan
for b in range(0, bins.size - 1):
if bins[b] <= val < bins[b + 1]:
the_bin_no = b
elif val > bins[bins.size - 1]:
the_bin_no = bins.size - 1
return the_bin_no
def calculate_pattern_poly_coeff(file_name):
y_source, sr_source = librosa.load(os.path.join(ROOT_DIR, file_name), sr=sr)
f0_source, voiced_flag, voiced_probs = librosa.pyin(y_source, fmin=librosa.note_to_hz('C2'),
fmax=librosa.note_to_hz('C7'), pad_mode='constant',
center=True, frame_length=4096, hop_length=512, sr=sr_source)
all_freq_bins = librosa.core.fft_frequencies(sr=sr, n_fft=n_fft)
f0_freq_bins = list(filter(lambda x: np.isfinite(x), map(lambda val: _get_bin_nr(val, all_freq_bins), f0_source)))
return np.polynomial.polynomial.polyfit(np.arange(0, len(f0_freq_bins), 1), f0_freq_bins, 3)
def calculate_pattern_poly_func(coefficients):
return np.poly1d(coefficients)
Method calculate_pattern_poly_coeff calculates polynomial coefficients.
Using pythons poly1d lib I can compute function which can modify the speech. How to do that?
I just need to move up or down all values vertically at certain point in time.
for instance I want to move all frequencies at time bin 0,75 seconds up 3 times -> it means that frequency will be increased and the melody at that point will sound higher.
Code:
def transform(sentence_audio_sample, mode=None, show_spectrograms=False, frames_from_end_to_transform=12):
# cutting out silence
y_trimmed, idx = librosa.effects.trim(sentence_audio_sample, top_db=60, frame_length=256, hop_length=64)
stft_original = librosa.stft(y_trimmed, hop_length=hop_length, pad_mode='constant', center=True)
stft_original_roll = stft_original.copy()
rolled = stft_original_roll.copy()
source_frames_count = np.shape(stft_original_roll)[1]
sentence_ending_first_frame = source_frames_count - frames_from_end_to_transform
sentence_len = np.shape(stft_original_roll)[1]
for i in range(sentence_ending_first_frame + 1, sentence_len):
if mode == 'question':
by = int(_question_pattern(i) / 500)
elif mode == 'exclamation':
by = int(_exclamation_pattern(i) / 500)
else:
by = 0
rolled = _roll_column(rolled, i, by)
transformed_data = librosa.istft(rolled, hop_length=hop_length, center=True)
def _roll_column(two_d_array, column, shift):
two_d_array[:, column] = np.roll(two_d_array[:, column], shift)
return two_d_array
In this case I am simply rolling up or down frequencies referencing certain time bin.
This needs to be polished as it doesn't take into consideration an actual state of the transformed sample. It just rolls it up/down according to the factor calculated using the polynomial function computer earlier.
You can check full code of my project at github, "audio" package contains pattern calculator and audio transform algorithm described above.
Feel free to ask if something's unclear :)

Plotting trajectories in python using matplotlib

I'm having some trouble using matplotlib to plot the path of something.
Here's a basic version of the type of thing I'm doing.
Essentially, I'm seeing if the value breaks a certain threshold (6 in this case) at any point during the path and then doing something with it later on.
Now, I have 3 lists set-up. The end_vector will be based on the other two lists. If the value breaks past 2 any time during a single simulation, I will add the last position of the object to my end_vector
trajectories_vect is something I want to keep track of my trajectories for all 5 simulations, by keeping a list of lists. I'll clarify this below. And, timestep_vect stores the path for a single simulation.
from random import gauss
from matplotlib import pyplot as plt
import numpy as np
starting_val = 5
T = 1 #1 year
delta_t = .1 #time-step
N = int(T/delta_t) #how many points on the path looked at
trials = 5 #number of simulations
#main iterative loop
end_vect = []
trajectories_vect = []
for k in xrange(trials):
s_j = starting_val
timestep_vect = []
for j in xrange(N-1):
xi = gauss(0,1.0)
s_j *= xi
timestep_vect.append(s_j)
trajectories_vect.append(timestep_vect)
if max(timestep_vect) > 5:
end_vect.append(timestep_vect[-1])
else:
end_vect.append(0)
Okay, at this part if I print my trajectories, I get something like this (I only posted two simulations, instead of the full 5):
[[ -3.61689976e+00 2.85839230e+00 -1.59673115e+00 6.22743522e-01
1.95127718e-02 -1.72827152e-02 1.79295788e-02 4.26807446e-02
-4.06175288e-02] [ 4.29119818e-01 4.50321728e-01 -7.62901016e-01
-8.31124346e-02 -6.40330554e-03 1.28172906e-02 -1.91664737e-02
-8.29173982e-03 4.03917926e-03]]
This is good and what I want to happen.
Now, my problem is that I don't know how to plot my path (y-axis) against my time (x-axis) properly.
First, I want to put my data into numpy arrays because I'll need to use them later on to compute some statistics and other things which from experience numpy makes very easy.
#creating numpy arrays from list
#might need to use this with matplotlib somehow
np_trajectories = np.array(trajectories_vect)
time_array = np.arange(1,10)
Here's the crux of the issue though. When i'm putting my trajectories (y-axis) into matplotlib, it's not treating each "list" (row in numpy) as one path. Instead of getting 5 paths for 5 simulations, I am getting 9 paths for 5 simulations. I believe I am inputing stuff wrong hence it is using the 9 time intervals in the wrong way.
#matplotlib stuff
plt.plot(np_trajectories)
plt.xlabel('timestep')
plt.ylabel('trajectories')
plt.show()
Here's the image produced:
Obviously, this is wrong for the aforementioned reason. Instead, I want to have 5 paths based on the 5 lists (rows) in my trajectories. I seem to understand what the problem is but don't know how to go about fixing it.
Thanks in advance for the help.
When you call np_trajectories = np.array(trajectories_vect), your list of trajectories is transformed into a 2d numpy array. The information about its dimensions is stored in np_trajectories.shape, and, in your case, is (5, 9). Therefore, when you pass np_trajectories to plt.plot(), the plotting library assumes that the y-values are stored in the first dimension, while the second dimension describes individual lines to plot.
In your case, all you need to do is to transpose your np_trajectories array. In numpy, it is as simple as
plt.plot(np_trajectories.T)
plt.xlabel('timestep')
plt.ylabel('trajectories')
plt.show()
If you want to plot the x-axis as time, instead of steps of one, you have to define your time progression as a list or an array. In numpy, you can do something like
times = np.linspace(0, T, N-1)
plt.plot(times, np_trajectories.T)
plt.xlabel('timestep')
plt.ylabel('trajectories')
plt.show()
which produces the following figure:

Using adaptive time step for scipy.integrate.ode when solving ODE systems

I have to just read Using adaptive step sizes with scipy.integrate.ode and the accepted solution to that problem, and have even reproduced the results by copy-and-paste in my Python interpreter.
My problem is that when I try and adapt the solution code to my own code I only get flat lines.
My code is as follows:
from scipy.integrate import ode
from matplotlib.pyplot import plot, show
initials = [1,1,1,1,1]
integration_range = (0, 100)
f = lambda t,y: [1.0*y[0]*y[1], -1.0*y[0]*y[1], 1.0*y[2]*y[3] - 1.0*y[2], -1.0*y[2]*y[3], 1.0*y[2], ]
y_solutions = []
t_solutions = []
def solution_getter(t,y):
t_solutions.append(t)
y_solutions.append(y)
backend = "dopri5"
ode_solver = ode(f).set_integrator(backend)
ode_solver.set_solout(solution_getter)
ode_solver.set_initial_value(y=initials, t=0)
ode_solver.integrate(integration_range[1])
plot(t_solutions,y_solutions)
show()
And the plot it yields:
In the line
y_solutions.append(y)
you think that you are appending the current vector. What actally happens is that you are appending the object reference to y. Since apparently the integrator reuses the vector y during the integration loop, you are always appending the same object reference. Thus at the end, each position of the list is filled by the same reference pointing to the vector of the last state of y.
Long story short: replace with
y_solutions.append(y.copy())
and everything is fine.

Matplotlib Magnitude_spectrum Units in Python for Comparing Guitar Strings

I'm using matplotlib's magnitude_spectrum to compare the tonal characteristics of guitar strings. Magnitude_spectrum shows the y axis as having units of "Magnitude (energy)". I use two different 'processes' to compare the FFT. Process 2 (for lack of a better description) is much easier to interpret- code & graphs below
My questions are:
In terms of units, what does "Magnitude (energy)" mean and how does it relate to dB?
Using #Process 2 (see code & graphs below), what type of units am I looking at, dB?
If #Process 2 is not dB, then what is the best way to scale it to dB?
My code below (simplified) shows an example of what I'm talking about/looking at.
import numpy as np
from scipy.io.wavfile import read
from pylab import plot
from pylab import plot, psd, magnitude_spectrum
import matplotlib.pyplot as plt
#Hello Signal!!!
(fs, x) = read('C:\Desktop\Spectral Work\EB_AB_1_2.wav')
#Remove silence out of beginning of signal with threshold of 1000
def indices(a, func):
#This allows to use the lambda function for equivalent of find() in matlab
return [i for (i, val) in enumerate(a) if func(val)]
#Make the signal smaller so it uses less resources
x_tiny = x[0:100000]
#threshold is 1000, 0 is calling the first index greater than 1000
thresh = indices(x_tiny, lambda y: y > 1000)[1]
# backs signal up 20 bins, so to not ignore the initial pluck sound...
thresh_start = thresh-20
#starts at threshstart ends at end of signal (-1 is just a referencing thing)
analysis_signal = x[thresh_start-1:]
#Split signal so it is 1 second long
one_sec = 1*fs
onesec = x[thresh_start-1:one_sec+thresh_start-1]
#process 1
(spectrum, freqs, _) = magnitude_spectrum(onesec, Fs=fs)
#process 2
spectrum1 = spectrum/len(spectrum)
I don't know how to bulk process on multiple .wav files so I run this code separately on a whole bunch of different .wav files and i put them into excel to compare. But for the sake of not looking at ugly graphs, I graphed it in Python. Here's what #process1 and #process2 look like when graphed:
Process 1
Process 2
Magnetude is just the absolute value of the frequency spectrum. As you have labelled in Process 1 "Energy" is a good way to think about it.
Both Process 1 and Process 2 are in the same units. The only difference is that the values in Process 2 has been divided by the total length of the array (a scalar, hence no change of units). Normally this happens as part of the FFT, but sometimes it does not (e.g. numpy.FFT doesn't include the divide by length).
The easiest way to scale it to dB is:
(spectrum, freqs, _) = magnitude_spectrum(onesec, Fs=fs, scale='dB')
If you wanted to do this yourself then you would need to do something like:
spectrum2 = 20*numpy.log10(spectrum)
**It is worth noting that I'm not sure if you should be applying the /len(spectrum) or not. I would suggest using the scale='dB' !!
To convert to dB, take the log of any non-zero spectrum magnitudes, and scale (scale to match a calibrated mic and sound source if available, or use an arbitrarily scale to make the levels look familiar otherwise), before plotting.
For zero magnitude values, perhaps just replace or clamp the log with whatever you want to be on the bottom of your log plot (certainly not negative-infinity).

Moving average of an array in Python

I have an array where discreet sinewave values are recorded and stored. I want to find the max and min of the waveform. Since the sinewave data is recorded voltages using a DAQ, there will be some noise, so I want to do a weighted average. Assuming self.yArray contains my sinewave values, here is my code so far:
filterarray = []
filtersize = 2
length = len(self.yArray)
for x in range (0, length-(filtersize+1)):
for y in range (0,filtersize):
summation = sum(self.yArray[x+y])
ave = summation/filtersize
filterarray.append(ave)
My issue seems to be in the second for loop, where depending on my averaging window size (filtersize), I want to sum up the values in the window to take the average of them. I receive an error saying:
summation = sum(self.yArray[x+y])
TypeError: 'float' object is not iterable
I am an EE with very little experience in programming, so any help would be greatly appreciated!
The other answers correctly describe your error, but this type of problem really calls out for using numpy. Numpy will run faster, be more memory efficient, and is more expressive and convenient for this type of problem. Here's an example:
import numpy as np
import matplotlib.pyplot as plt
# make a sine wave with noise
times = np.arange(0, 10*np.pi, .01)
noise = .1*np.random.ranf(len(times))
wfm = np.sin(times) + noise
# smoothing it with a running average in one line using a convolution
# using a convolution, you could also easily smooth with other filters
# like a Gaussian, etc.
n_ave = 20
smoothed = np.convolve(wfm, np.ones(n_ave)/n_ave, mode='same')
plt.plot(times, wfm, times, -.5+smoothed)
plt.show()
If you don't want to use numpy, it should also be noted that there's a logical error in your program that results in the TypeError. The problem is that in the line
summation = sum(self.yArray[x+y])
you're using sum within the loop where your also calculating the sum. So either you need to use sum without the loop, or loop through the array and add up all the elements, but not both (and it's doing both, ie, applying sum to the indexed array element, that leads to the error in the first place). That is, here are two solutions:
filterarray = []
filtersize = 2
length = len(self.yArray)
for x in range (0, length-(filtersize+1)):
summation = sum(self.yArray[x:x+filtersize]) # sum over section of array
ave = summation/filtersize
filterarray.append(ave)
or
filterarray = []
filtersize = 2
length = len(self.yArray)
for x in range (0, length-(filtersize+1)):
summation = 0.
for y in range (0,filtersize):
summation = self.yArray[x+y]
ave = summation/filtersize
filterarray.append(ave)
self.yArray[x+y] is returning a single item out of the self.yArray list. If you are trying to get a subset of the yArray, you can use the slice operator instead:
summation = sum(self.yArray[x:y])
to return an iterable that the sum builtin can use.
A bit more information about python slices can be found here (scroll down to the "Sequences" section): http://docs.python.org/2/reference/datamodel.html#the-standard-type-hierarchy
You could use numpy, like:
import numpy
filtersize = 2
ysums = numpy.cumsum(numpy.array(self.yArray, dtype=float))
ylags = numpy.roll(ysums, filtersize)
ylags[0:filtersize] = 0.0
moving_avg = (ysums - ylags) / filtersize
Your original code attempts to call sum on the float value stored at yArray[x+y], where x+y is evaluating to some integer representing the index of that float value.
Try:
summation = sum(self.yArray[x:y])
Indeed numpy is the way to go. One of the nice features of python is list comprehensions, allowing you to do away with the typical nested for loop constructs. Here goes an example, for your particular problem...
import numpy as np
step=2
res=[np.sum(myarr[i:i+step],dtype=np.float)/step for i in range(len(myarr)-step+1)]

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