Harmonic | high performance and scalable RTMP | Video Utils library
kandi X-RAY | Harmonic Summary
kandi X-RAY | Harmonic Summary
A high performance RTMP live streaming application framework.
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Currently covering the most popular Java, JavaScript and Python libraries. See a Sample of Harmonic
Harmonic Key Features
Harmonic Examples and Code Snippets
def harmonic_mean(series: list) -> float:
"""
return the harmonic mean of series
>>> harmonic_mean([1, 4, 4])
2.0
>>> harmonic_mean([3, 6, 9, 12])
5.759999999999999
>>> harmonic_mean(4)
T
def harmonic_series(n_term: str) -> list:
"""Pure Python implementation of Harmonic Series algorithm
:param n_term: The last (nth) term of Harmonic Series
:return: The Harmonic Series starting from 1 to last (nth) term
Examples:
def harmonic(number: int) -> float:
"""Calculate the harmonic number of the given number.
Args:
number (int): input integer
Returns:
float: harmonic number
>>> harmonic(2)
1.5
>>> harmon
Community Discussions
Trending Discussions on Harmonic
QUESTION
I am using HealPix to calculate the outflow rate of particles through a sphere with some radius and I am trying to determine coefficients of spherical harmonics (basically fit some data with spherical harmonics).
The basic steps are:
...ANSWER
Answered 2022-Mar-25 at 08:58In order to get the same values of a_lm coefficients it is necessary to normalize values of individual pixels with the pixel surface - dividing with 4*pi/npix.
Another thing, in case it might be helpful to someone. The outflow rate can be easily computed from
QUESTION
Background : I've connected an ADXL335 accelerometer to pins A0,A1,A2 on the Arduino Uno. During the measurement, the Arduino is connected to the laptop via usb, and accelerometer is powered directly by the 3.3V pin on the Arduino. The intention is to read voltage from the accel and plot the corresponding time and frequency spectrum using Python. The overall process is quite straightforward and I'm actually able to achieve this. A snapshot of a signal I recorded is shown below. Arduino code to read/write from the accelerometer and Python code to read from Arduino and construct the plots are provided at the bottom for reference.
The issue I'm facing is with the actual values read by the Arduino (I think).
Problem: There seems to be some noise picked up by the Arduino during the read/write process. To demonstrate this, I ran two separate data acquisitions with the sensor at rest. Snapshots are provided below. I had used a similar system in 7yrs ago in grad school, using National Instruments DAC and LabView. I did experience some noise back then but that was akin to electrical noise ~60 Hz. This one is quite peculiar and I don't know how to deal with it.
Closing: Could someone please help me understand what noise it is, that I'm seeing? Perhaps knowing what this is called will help me research about it in detail and try to tackle it. Any pointers/references would help as well.
Future Scope: I believe that a stable at-rest signal as the first step will greatly help mitigate the offset in Voltage (DC-component in freq-spectrum) even with a simple average. Also, I need to learn how to increase the sampling rate from the uno.
Programs used:
Arduino code to read/write from the accelerometer
...ANSWER
Answered 2022-Mar-22 at 21:24This is spike noise. You should use a median filter to clean your signal. It is a digital non linear filter which you can implement in your software.
Just a note: I don't know which sample frequency are you using but the higher is the sample frequency, the more is the collected noise. So, limit the sample frequency to 2 x signal bandwidth (Shannon theorem)
QUESTION
I'm not seeing the mathematical results I'm expecting from the following code, which I believe should produce the Harmonic Series:
...ANSWER
Answered 2022-Mar-17 at 09:32You run Fraction(x)
where x
is a float. This is too late, you already lost precision, so your fraction's precision is as good as that of the float.
Use Fraction
in the function:
QUESTION
I'm trying to setup a pipeline allowing me to detect musical notes from audio samples, but the input layer where I identify the frequency content of the samples does not land on the expected values. In the example below I...
- build what I expect to be a 440Hz (A4) sine wave in the FFTW input buffer
- apply the Hamming window function
- lookup the first half the output bins to find the 4 top values and their frequency
ANSWER
Answered 2022-Mar-14 at 09:45I think you've misunderstood the M_2_PI
constant in your GenerateSinWave
function. M_2_PI
is defined as 2.0 / PI
.
You should be using 2 * M_PI
instead.
This mistake will mean that your generated signal has a frequency of only around 45 Hz. This should be close to the output frequencies you are seeing.
The same constant needs correcting in your HammingWindow
function too.
QUESTION
I've been working on a Tone.js synthesizer project for some time now.
For the record I will include the links: Repo Deployment (it is still under development as I am stuck with this issue)
I have encountered a serious issue that I couldn't manage to solve. I use the PolySynth with an FMSynth, but I've had the same issue with all the other synth types that I tried.
Whenever I am trying to tweak the parameters of the synth live (ex. detune, modulation index, amplitude envelope etc.) I get unexpected behaviour:
Most of the times, when I change the value of a parameter, it works at once, thus the sound gets modifidied according to the change I made, but the sound is then stuck to the first modified value, even if I keep changing the value of the parameter.
Then sometimes I get the modified sound every second time I play a note on the synth. One time I get the modified sound and then next time the original sound without any modification, then the modified sound again and so on.
Somestimes it works, but I am still stuck on the first modification.
Sometimes it works randomly, after playing some notes first.
Sometimes it works at once, but then some specific notes produce the unmodified original sound, regardless of my modification (and still the synth stops responding to any further parameter changes).
This is happening with every parameter but volume: volume works as intended every time.
Let's use Modulation index as an example (the same happens with detune, harmonicity and attack - those are the parameters I've implemented for the time being). Originally, I use NexusUI components, but for this I wil be using a regular HTML slider (to prove that NexusUI is not the problem). It can be found in the deployment website I provided and in the repo. This is my code:
In the main JavaScript file, I create the synth and send it to destination:
...ANSWER
Answered 2022-Mar-04 at 12:23With the Tone.js instruments, use the .set
method for changing properties. You can change multiple properties at the same time, like this:
QUESTION
I am currently writing a program to calculate the Lucas Series, Hexagonal Series, and Harmonic Series given a number from an input file. I am using a pipe and shared memory segment to share values across the programs.
I start by creating my pipe
...ANSWER
Answered 2022-Feb-24 at 23:32Your problem is that after doing shm_open
, doing fstat
on the return value shows an st_size
value of 0!
Initially, the descriptor from shm_open
has zero size. It must be explicitly given a size. Note that just specifying 32 as a length to mmap
does not cause the file to be expanded.
This must be done with ftruncate
.
After the shm_open
call, you need to do:
QUESTION
The following code I have written measures PSD of the signal y by taking FFT first.
...ANSWER
Answered 2022-Feb-21 at 11:35You can normalize the PSD, thus the maximum of the PSD will be 0dB (if some harmonic is higher than the signal you should adapt the value to normalize):
QUESTION
Trying to create some musical notes by combining harmonic series. Very simple code, but the audio turns up blank. Any thoughts?
...ANSWER
Answered 2022-Jan-24 at 04:00The sound generated by the code is audible but weak.
I have no experience in audio programming, but some type of noise resembling a loud beep can be generated by the following:
QUESTION
I am trying to find peaks in spectrum but I need to extract only peak of base frequency and its harmonics, red rectangular. How to exclude anything before base frequency and only include base frequency and its 3 harmonics. I use this code but it does not help. Any idea?
...ANSWER
Answered 2021-Dec-16 at 18:59EDITED to adapt to new code with an example. We are applying on the envelope spectrum of a signal - [ES,F] = envspectrum(sig,Fs);
-, so we know the signal and its frequency sampling (fs).
Still being the same process. You can just compute the values in findpeaks()
for only those samples of your signal which are higher than 250 Hz. For that purpose, you define a logical array for the frequencies values you want to:
QUESTION
I am leveraging code from an existing answer. In that answer the signal oscillates around zero. In my data, I am shifted above the x axis however the curve fitting algorithm does not capture that effectively.
...ANSWER
Answered 2021-Nov-30 at 21:37You have problem with first coefficient (a[0]
), constant. You should not multiply it by cosine. Take a look at wiki.
I am not sure if this scaling (dividing by 2) is necessary in your case, but it is good to follow some conventions:
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