muno17 / neblinita

sonic fog machine

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neblinita - sonic fog machine

As a lifelong musician, I have always been intrigued with audio effects such as delays, echos and distortion and how they're able to transform any sound into something new. In particular, I'm quite fond of reverbs and their ability to transport the listener into an immersive space of sonic ambiance.
For my final project in CS50P, I wanted to use my newly acquired skill of coding in Python to create something that I could use in my daily audio explorations; thus, neblinita was born.

neblinita is an audio processing unit that adds texture and ambiance to an input audio signal. The basis of neblinita is two flavor of reverbs I created, luz and sombra, which are mixed together with the input signal to create a spacious, beautifully distorted atmosphere:

luz creates harmonic content and a large atmosphere by sending a copy of the input signal through an initial delay which was designed to add echos at very short intervals allowing the sound to be drawn out but without the usual noticeable separation in echos found in a traditional delay. This signal is then fed into a more traditional delay featuring a large amount of feedback which is separated in lengthy intervals. The combination of feeding the first delay into the second delay creates a smooth signal that is then multiplied and repeated over time, creating modulation and adding variance to the signal over time. This signal is then fed into a chorus which adds color by adding harmonic content to the signal. Since the delayed signal feeding into the chorus is already being modulated with each repeat, the chorus helps create a massive field of harmonic content since the delay signal is changing in frequency over time. This new signal is then finally fed into a large reverb. A reverb is essentially a large amount of delays repeating very rapidly to give the impression of a sound echoing in a given space. The larger the time of the reverb, the larger the space that is created. Feeding an already long, heavily delayed signal into a reverb with a long time creates an incredibly large and deep atmosphere. The output of luz is therefore a very atmospheric, harmonic, psychedelic signal.

sombra creates a gritty, chaotic ambiance by heavy use of distortion. A copy of the input signal is sent through a distortion, overdriving the sound and creating a fuzzy, dirty output. This is then sent into a delay similar to the first delay found in *luz which smoothens out the sound and adds length without an audible separation in the sound. Before the sound is output of this initial delay, the signal is blasted through another distortion, this time being controlled by the user. This gnarly signal is then ran through a more traditional delay, adding repeats and chaos to the gnar gnar. This is finally fed into a reverb to add a large amount of space and create a dirty, nasty signal giving the listener the impression of getting sent into The Void.

The initial input, also known as the dry signal, is then send to a mixer along with the signal output by luz and sombra. These signals are then mixed together to create the final signal to be output. A 'wet/dry' feature was implemented to allow the user to control the amount that the signals should be mixed. When the signal is fully dry, all you hear is the original dry signal. When the signal is fully wet, you only hear the outputs of luz and sombra. Setting the wet/dry anywhere in between fully wet and fully dry will attenuate each output accordingly, allowing the user to mix the dry signal with the effects of neblinita as they see fit.


I used the Pyo library to handle all of the audio routing:

Within the project folder is a program I created named 'audio_io.py'. Running this program before starting the main program is essential to assure that your audio will be routed correctly and to ensure that audio feedback loops won't be created (trust me you do not want this to happen for the sake of your ears). audio_io.py will output information pertaining to the audio inputs and outputs of your computer. In particular, we want to pay special attention to the 'default input' and 'default output' at the bottom. You will want these to match the desired devices listed in the 'AUDIO devices' section above. For example, I want to input signal into my computer through my Zoom H4N Pro recorder and be able to listen through my headphones that I have plugged into my computer. When I run audio_io.py, my 'AUDIO devices' and default i/o are listed as:

0: OUT, name: H4, host api index: 0, default sr: 48000 Hz, latency: 0.004354 s

1: IN, name: H4, host api index: 0, default sr: 44100 Hz, latency: 0.005896 s

2: OUT, name: External Headphones, host api index: 0, default sr: 44100 Hz, latency: 0.006213 s

3: IN, name: MacBook Pro Microphone, host api index: 0, default sr: 48000 Hz, latency: 0.033687 s

4: OUT, name: MacBook Pro Speakers, host api index: 0, default sr: 44100 Hz, latency: 0.009229 s

5: IN, name: Microsoft Teams Audio, host api index: 0, default sr: 48000 Hz, latency: 0.010000 s

5: OUT, name: Microsoft Teams Audio, host api index: 0, default sr: 48000 Hz, latency: 0.001333 s

default input: 1

default output: 2

In this example, the defaults are already set for where I would like to input and output the audio. In the main Project.py file I would then go to line 22 and enter in the correct input and output numbers, 1 for the input and 2 for the output in this case:

s.setInputDevice(1)

s.setOutputDevice(2)

!!!!!!!!! WARNING !!!!!!!!! MAKE SURE THAT THE INPUT AND OUTPUT COMBINATION IS NEVER INPUT - INTERNAL MICROPHONE (3 in the example above) AND OUTPUT - INTERNAL SPEAKERS (4 in the example above). THIS WILL CREATE AN AUDIO FEEDBACK LOOP SINCE THE BUILD IN MICROPHONE WILL PICK UP ANYTHING COMING OUT OF THE INTERNAL SPEAKERS, CREATING AN INFINITE LOOP OF EVER EXPANDING PAIN TO YOUR EARS. IF YOU DO THIS BY ACCIDENT THEN EXIT THE PROGRAM IMMEDIATELY TO AVOID EAR DAMAGE !!!!!!!!!


I also used the Pyo library for the sound engine.

I knew very little about digital signal processing (DSP) going into this project so I thoroughly read through all of the Pyo documentation to try and get a grasp on the subject, or least enough to be able to build this project. The general concept is that pyo creates a 'server' where all the audio runs through. Within this server, you run operations on 'pyo objects' which are essentially classes pertaining to each component of an audio system. neblina was designed to take in an audio signal and then create two copies of that signal. Each signal is then fed through the processes described in the first segment of this ReadMe.

The pyo documentation lists effect types in two ways. The first is in a simple function call, such as Disto() to apply distortion to an effect. These functions seemed tempting to use because of their ease of use but they don't allow for much customization. The second is by presenting broken down bare-bones functions of different effects. These templates (for example, http://ajaxsoundstudio.com/pyodoc/examples/07-effects/03-fuzz-disto.html for distortion) show you the basic building blocks that make up a certain type of effect. From these templates you're then able to make your own modifications and add/remove functions as you see fit. I took this latter approach for all of my effects since I wanted something fully customizable to my liking that would sound like the idea I had in my head when I first thought of making this project. This of course took a lot of experimentation since I'm a DSP noob but, after a lot of trial and error, I finished customizing all of my effects exactly how I wanted them to function. After creating these effects, I chained them all together by creating return values (pyo objects) for each effect which I would then assign and then send through the next effect. After running through each chain, the final reverb effect of both luz and sombra returns a pyo object containing the final output of each signal chain.

The pyo objects for luz and sombra are then summed in a Mixer object along with the initial dry signal. This again took a lot of experimentation to get the output levels of each effect to a proper level so as to not output a signal so loud that it would cause audio dropouts and damage speakers/headphones. The method I used was to attenuate each signal with a float value between 0.0 and 1.0. The output for the input signal combined with luz and sombra would then max out between -15 to -5 db depending on the level of the initial signal. This of course doesn't factor in devious individuals who are out to make speakers and eardrums explode by sending very loud input signal but in this case, and the way the majority of virtual effects processors work, we assume the user is a rational well-intended person. I also created a wet/dry function that calculates new attenuation levels based on how wet or dry the user wants the signal to be. This is all then output and the result is a glorious wave of distorted space and sound.


I created a simple GUI using the PySimpleGui library.

I initially attempted to create a GUI that would let you control eight different parameters in neblinita but I quickly realized that this would complicate things in a very sonically-ugly manner. Throughout all the research that was conducted for this project, one common trend I kept seeing was people having this whole "don't use python for audio processing, since it's a higher level language it's not comparatively as fast and won't be able to handle audio as well as a lower level language such as C or C++" mentality. I initially thought this was maybe overexagerrated by people out to bash Python. Before implementing a GUI, neblinita sounded exactly how I wanted it to and there didn't seem to be any issues with processing the audio. However, once I started creating modifiable parameters through the use of a GUI, I quickly realized that maybe there was a lot of truth into what I kept reading about C++ superiority in audio processing.

The first GUI element I added was a slider to be able to control the wet/dry level since I think that's the single most crucial part of any effects processor since you want to let the user be able to control how much of an effect is added to their signal. After hours(days) of trying to figure out how to implement this through a GUI, I finally succeeded! Or, at least I thought I had. I had a slider that would definitely do what it was intended to do but the issue now was that if I was sending audio through the effect and changing the wet/dry parameter at the same time then I would experience a lot of crackling, hissing and audio dropouts which made the sound not sound at all how I intended. I thought this was maybe something small that I could fix later, so I kept on more GUI elements. I had slider that could now control the delay time and the amount of space created by luz and sombra! But now the issue kept getting worse and worse with more unintended audio artifacts ruining my once beautiful sound :(. I thought maybe I would add the ability to change the amount of distortion added to sombra and that that would somehow fix everything but boy was I wrong.

At this point I had a functioning GUI but when I tried to do anything with it my sound would get mangled and violently sent back out as the worst thing that has ever gone into your ear holes. Then I realized why this was happening and why maybe(probably) Python is not the best language to use for audio processing. Much like an image, sound is interpretted the by computers as tiny slices that together make up the whole: samples are the pixels of the audio world. In order for a computer to generate a signal that can be interpreted correctly by a human ear, the amount of samples that need to be churned out per second is typically at least 44100. If you look through the code of my project, you'll see that samples are what are being sent through everything. Calculations are then ran on the samples at a rate of 44100 per second, or one every 0.005804988662131519 seconds, and this process continues endlessly through every calculation as long as an input signal is being received. That is a lot of math going on every second. Now factor in changing values (sliders on the GUI) and now you have a lot of math having to be continually recalculated every time a slider is moved even a fraction (1/100 to be precise) and what you end up with is like seriously so much math. Since so much math is being calculated every second, Python then has to compile everything down to C until it is finally compiled down to binary for the computer to interpret. And then all of that has to continue being calculated nonstop, overloading the computer with information. In short, since the processes cannot be ran at a fast enough rate, my GUI was making it so that sound couldn't smoothly change with every adjustment to any of the parameters I had created.

To partially solve this, I dumbed my GUI down to be able to control two things: the wet/dry signal and the amount of distortion (labeled fog)

neblinita

The original name for this project was neblina but with this new, smaller GUI *neblinita was born.

Unfortunately, there are still some audio dropouts as you move the sliders to change either of the two parameters, but it's not nearly as bad as the original version I had made. My fix for now is to just simply quit sending an input signal, change the settings on the sliders once there's no sound, play something, and repeat the process until I get bored. I think this is an issue that I could fix in the future once my DSP and Python knowledge have improved, but for now I'm quite proud of this little unit.


Audio programming is something that I'm going to explore more since it combines my new hobby with my old one and allows me to create something that, for me, is very pleasing and useful. CS50P has given me the tools to start experimenting and creating projects that excite me. In the future I'm planning on creating a hardware unit neblina using neblinita as the prototype. I don't know how I'm going to do this at all but if CS50 has taught me anything it's that if I really want to, I can learn to do anything.


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sonic fog machine


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