Realtime audio manipulation - audio

Here is what i like to achieve:
I like to play around in creating "new" software / hardware instruments.
Sound processing and creation is always managed by software. But one could play the instrument via ultrasonic distance sensor for example. Another idea is to start playback when someone interrupts the light of a photoelectric barrier and so on....
So the instrument would play common sounds, but has to be used in an unusal way. For example, the ultrasonic instrument would play a sound if it detects something in a certain distance. The sound could be manipiulated in pitch for example if the distance gets smaller.
Basically i like to playback a sound sample and manipualte this in realtime.
I guess i have to use WAV samples for this, right? And which programming language do you think fits best for this task?
Edited after kevins hint: please kick me into the right direction - give me a hint where to start.
Thanks in advance

Since you're using the the Processing tag, you can try Processing.
It comes with a sound library like Minim or you can install beads which is great. There's actually a nice book on it: Sonifying Processing
You might find SuperColider fun as well.
The main thing is what are you comfortable with at the moment ?
If Processing syntax looks intimidating, you can actually try a different programming paradigm like data flow. In which case you can use PureData(free, opensource) or MaxMSP(very similar, but commercial). The idea is rather than typing instructions, you connect boxes with wires which is fun and the examples are great too.
If you're into c++ there are plenty of libraries. On the creative side, there's a nice set of libraries called OpenFrameworks that's easy and fun to use. If this is your cup of tea, have a peek at Maximilian.
Bottomline is: there are multiple options to achieve the same task. Choose the best tool for your (based on your background) or try each and see what you like best.

You asked "And which programming language do you think fits best for this task?" - I would also suggest using Processing. I have been used Processing to work with sounds previously. And in all cases I used Minim. It has many UgenS to generate sounds programmatically.
Also, you wants to integrate with some sensors. I'm not sure what types of sensors you will use, but Processing goes pretty well with different Arduino modules and sensors. Check this link for more direction.
Furthermore, you can export your project as .exe or executable .jar files. And their JS version (P5.js) works almost the same as the Java version.

Related

APCS final project: Converting an audio file to a simpler MIDI file

Lets say I have the audio file for Happy Birthday. I want to convert that audio file into an audio file that sounds like this : happy birthday.
First, I'd like to know if I have the ability to program this? Can a highschooler who's almost finished with APCS program this?
If I can:
How would I change the bpm of the song? I've searched through a bunch of websites, but they weren't very helpful.
I know that audio files can be represented in waveforms. How would I scan for each individual wave in an audio file (I need this to isolate the notes)?
This is a very ambitious project, actually. One reason is that it involves using digital signal processing tools like FFT (Fast fourier transforms) to analyze the sound to pick out the pitches. You might be able to find a library that can do this, but as far as coding such a tool, that would involve a steep learning curve.
If you would like to look further into this, there is a good online resource called "The Scientists and Engineers Guide to Digital Signal Processing". I was able to work through and understand the discrete fourier transform with only high school math (lots of trig) and a bit of calculus. It was a lift, though.
Trying to analyze rhythm is also no easy task. Even with advanced tools provided in professional notation system such as Finale, people have trouble playing rhythms in time well enough for the best transcription tools. Algorithms that "quantize" the beats help but also limit the amount of detail that can be included in the playback.
My guess is that as interesting and worthwhile as this project would be, to bring it to completion before the semester ends would require putting together prebuilt pieces. A lot of programming is done that way, these days.
If you scale the project back to something like just getting your code to analyze a short sample of a single note and give its pitch, that would be both impressive and doable with a lot of work. It could be done with a DFT algorithm instead of requiring FFT, reducing the amount of info you'd have to acquire first. That way, you'd only have to work your way up to understanding and implementing the material on this link which is about calculating the DFT. Notice that there is example code in BASIC. The code examples throughout this book are a big help.

Libgdx music/sound effect with reverb

Is it possible to add specific reverb to my sound effect/music track in libgdx?
I want to add outdoor/indoor reverb to make all tracks sounded the same.
I don't think that Libgdx has a mechanism to adding effects to sound. The Sound class delivers no function for this.
I see three solutions here:
Prepare two kinds of sounds (one with reverb one without - it is easy to do using software like Audacity - and play one or another due to environment of player's current being
Try to implement it yourself
I see that in the Sound class there is setPitch(long soundId, float pitch) method. Due to Wikipedia the reverb is just a kind of echo so maybe (but not for sure) you could achieve the effect by
making copy of sound
slowering it a little
lowering the volume
playing simultaneously with original sound
Find 3rd part library that will do it for you - the Google returns some examples of libs working with libgdx like SoundTouch Audio Processing Library - maybe you will find something usefull
First one is the easiest and if you are not afraid of space problems I would strongly recommend it to you (althought why not to try implementing it)
I've implemented reverb, positional audio, and arbitrary filters using OpenAL against the latest libgdx 1.10+/lwjgl 3+ with this demo code, based off of gdx-sfx (which only works with lwjgl 2) and libgdx-audio-effects.
I'd like to promote this into a fully fledged library at some point 😂

Choosing an audio API

I'm struggling to choose between a vast number of audio programming languages and APIs. I'm very (totally) new to audio programming so please bear with me.
Software
I need to be able to:
Alter volume of different sounds before outputting them to anything (these sounds can have a variety of different origins, for example mp3s and microphone input)
phase shift sounds
superimpose sounds that I have tweaked (as per items 1 and 2)
control the output to each of 8 channels independently of one another
make this all happen on Windows7
These capabilities need be abstracted by a graphical frontend I will probably make myself. What I want to be able to do is create 'sound sources' and move them around a 3D environment along either pre-defined trajectories and/or in relation to the movement of whoever is inside the rig. The reason I want to do pitch bending is so I can mess with red-shift stuff.
I don't want to have to construct full tracks before-hand and just play them. I want the sound that is played to depend on external input from sensors as well as what I am doing on the frontend.
As far as I know this means I cant use any existing full audio making app.
The Question
I've been looking around for for the API or language I should use and I have not turned up a blank, quite the opposite actually. I'm struggling to narrow down my search. A lot of my problem stems from the fact that I have no experience in audio programming.
So, does anyone know off-hand of an API or language that meets my criteria?
Hardware stuff and goals
(I left this until last because I'm not sure how relevant it is)
My goal is to make three rings of speakers at different heights and to have enough control over them to be able to simulate any number of 'sound sources' within the array. The idea is to have someone stand in the middle of the rig and be able to make it sound like there are lots of things moving around them. To get this working I'm planning on doing a little trig and using 8 channels of audio from my PC. The maths is pretty straight forward, it just the rest that I need to worry about
What I want to do next is attach a bunch of cameras to the thing and do some simple image recognition stuff to be able to 'attach sound sources' to different objects. Eg. If someone is standing in the right place it can be made to seem as though all red balls quack like a duck, and all orange ones moan hauntingly.
This is not to detract from Richard Small's answer, but to comment on some of the other options out there:
If you are looking for something higher-level with which you can prototype and develop this faster, you want max/msp or it's open source competitor puredata. These are designed for musicians who are technically minded, but not so much for programmers. As a result, you can build this sort of thing quickly and efficiently.
You also have some lower level options: PortAudio can handle your audio I/O, you would have to do the sound generation and effects and so on on your own or with other libraries. Cinder and OpenFramewoks both provide interfaces for audio, cameras, and other stuff for "creative programming". I'm afraid I don't know if they meet your full requirements, but they are powerful and popular for this sort of thing so I encorage you to look at them.
The two major ones these days tend to be
WWise
WWise Download Link
FMOD
FMOD Download Link
These two engines may even in fact be overkill for what you need, but I can almost guarantee that they will be capable of anything you require.

Audio support for programming languages

I want to start on a hobby project that focuses on displaying audio files in a folder in a certain fashion and has the ability to play such an audio file and shows basic control options for playing. However, i'm struggling to find a fit programming language for this.
The displaying part shouldn't be too hard and can probably be done in most of the programming languages. The audio part is what concerns me the most since it's not the main focus of the project and should only do limited things (so it shouldn't be too hard) and i do not know anything about sound support in the programming languages i currently know. (Java, C and C++)
Specifically i would like to be able to do these things:
Play a sound file
Stop/pause a playing song
Adjust volume
Show a bar that displays the current position in the song
Most files will be .mp3 files but being able to process other formats is certainly a plus. Since this is just a small project it's ok if it runs just on Windows. Scalabilty would be nice but not required.
It would be nice to have a small overview of audio support/audio libraries of programming languages (i'm always up for something new) that can accomplish these simple things, in a not too complicated way, aswell as personal experiences.
In this way i hope to create a better understanding of which programming language fits my project best. (i would very much like to not have to change language mid-way the project)
--
Edit:
This is only for a later stage of the project if the first part was successfull: i will want to change the file names of the audio files that are displayed. (to make them follow a specific format)
I haven't written audio processing programs much, but I know a lot of them exist for C and C++. For Java perhaps, too, but I don't know Java. I had used audio with SDL in a game, but that doesn't have that many features and I don't recommend it.
There's this question asking for a library in C, and there are a couple of similar questions that SO brings up on the side. You may want to take a look at those.
You would also need to look for a library that loads different file types. SDL at least, only opens .wav files, which I believe most of the playback libraries would support. For MP3, you will most likely need an additional library. I know Audacity uses LAME Mp3 so I'm guessing that should be good.
Some of the functionalities you want is also doable by yourself. For example, knowing the length of the music and the amount you have already read, you will know how far in the audio you are. Adjusting the volume is also a multiplication (in the simplest case) that you can do on the audio data if the library doesn't provide it.
A very good choice seems to be PortAudio which is used by Audacity, and also recommended in the accepted answer of the question I mentioned above.
I've done audio apps in both Java and C++. Java development goes way faster because it's a more powerful language and has garbage collection, but JavaSound is a pretty awful solution for audio. Of course, there are wrappers for FFMPEG and other stuff, so you can get a lot of things working. Here's an example of a Java audio app: http://www.indabamusic.com/help/mantis
OTOH, C++ gives you lots of control, low latency and wealth of libraries. (another answer mentioned Portaudio, which is, indeed, great.) But you will definitely find it also has a much longer development cycle.
You can certainly do everything you want to do with either language.

TI-99 speech effect?

I want to make a program that takes recorded speech and transforms it so it sounds like it's coming from a Texas TI-99. Do you have any good ideas and resources for how to go about that?
Most of those old speech synthesizers were build directly in-chip. Perhaps you could find a synthesizer that sounds like the chip, but if you really want the original sound, you would either have to simulate the chip (I don't know if it's a simple matter, perhaps the chip internals aren't published).
I only know because I burnt out a number of the Radio Shack speech synthesizer ICs before I managed to get a SP0256-AL2 working.
If you're more of a do-it yourself type guy, you need to find out which IC actually drove the speech synthesis in a TI-99, and then build the chip up on a bread board. That's what I was trying to do back then, and I managed to get the chip to speak, but lost patience after I fried my third chip due to a mis-wiring issue when I attempted to attach it to my PC's parallel port. I think this was the book I was using back then, but there's no cover art featured so it's hard to know for sure.
If you are familiar with how to use ROM images, there seems to be a gentleman that has managed to refeverse engineer the ROM image out of a SP0256-AL2. Look here for the image and the incredible granted permission to do the work and distribute the results.
You could start with open source that does something similar: Adding Robotic/Vocoder effect to your song using Audacity

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