I've searched the net but didn't found anything interesting. Maybe I'm doing something wrong.
I'm looking for sound synthesis API written on C, C++ or even Objective-C, which can synthesize different types of waves, effects are optional.
Here's a complete library/toolkit for FM (Frequency Modulation) synthesis:
link1
link2
If you have time to spare... creating simple sound synthesis from scratch is actually a fun endeavor. If you create a small buffer of 256 16 bit samples which represent either a sine. a sawtooth, block or pulse, you can copy these to a live audiobuffer (e.g. a small buffer (say 16kb)) which constantly loops. By staying ahead of the playposition, and constantly filling up the buffer with new values, you can create the soundoutput.
You can use the small buffers to combine these in interesting ways (simplest is just to add them together (additive synthesis)).
The frequency of the tone can be manipulated by using a bigger or smaller sampling step through the small buffers. Amplitude can be manipulated by scaling the samples before putting them into the output buffer.
Great fun experimenting with this!
If you have this step nailed, you can add more sophisticated effects like filters (low pass, high pass, etc) and effects (reverbs, echoes, etc)
R
Have you looked at the synthesis toolkit (STK)? It's in C++ (I don't think ObjC is the right language for audio synthesis, in fact audio units, Apple's own way of doing audio stuff, including generators/filters/effects... is in C++).
STK will run on Mac OS X, and iOS no problem (CoreAudio is supported), but will also run on Linux and Windows (Direct sound and ASIO), using RtAudio. It's really nicely done and lightweight, these guys have spent a lot of time thinking about it and it will definitely give you a big head start. It can handle loads of different audio file formats + midi (and hopefully OSC soon...).
There is also Create and CLAM which is huge, these include GUI components and many other things which you might or might not want. If you're only interested in doing sound synthesis I really recommend STK.
PortAudio is also a great C API that we used last semester in an audio programming course. It provides an audio callback...what more could you need!?
I haven't tried incorporating it with anything in Objective-C yet, but will report back when I do.
Writing audio synthesis algorithms in C/obj-C is quite difficult in my opinion. I would recommend writing your signal processing algorithms using PureData and then use ZenGarden or libpd to embed and interpret the pd patches in your app.
Another C++ library is nsound:
http://nsound.sourceforge.net
One can generate any kind of modulated signal using the Generator class or using the provided Sine class. Each time-step can have it's own instantaneous frequency and phase offset.
You can also experiment with the Python module to prototype your algorithm quickly, then implement in C++. It can produce pretty matplotlib plots from Python and even from C++!
Have you looked at CSound? It's an incredibly flexible audio generation platform, and can handle everything from simple waveform generation to FM synthesis and all kinds of filters. It also provides MIDI support, and you can extend it by writing custom opcodes. There's a full C API and several C++ APIs as well.
Related
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.
I'm currently researching an problem regarding DOA (direction of arrival) regression for an audio source, and need to generate training data in the form of audio signals of moving sound sources. In particular, I have the stationary sound files, and I need to simulate a source and microphone(s) with the distances between them changing to reflect movement.
Is there any software online that could potentially do the trick? I've looked into pyroomacoustics and VA as well as other potential libraries, but none of them seem to deal with moving audio sources, due to the difficulties in simulating the doppler effect.
If I were to write up my own simulation code for dealing with this, how difficult would it be? My use case would be an audio source and a microphone in some 2D landscape, both moving with their own velocities, where I would want to collect the recording from the microphone as an audio file.
Some speculation here on my part, as I have only dabbled with writing some aspects of what you are asking about and am not experienced with any particular libraries. Likelihood is good that something exists and will turn up.
That said, I wonder if it would be possible to use either the Unreal or Unity game engine. Both, as far as I can remember, grant the ability to load your own cues and support 3D including Doppler.
As far as writing your own, a lot depends on what you already know. With a single-point mike (as opposed to stereo) the pitch shifting involved is not that hard. There is a technique that involves stepping through the audio file's DSP data using linear interpolation for steps that lie in between the data points, which is considered to have sufficient fidelity for most purposes. Lot's of trig, too, to track the changes in velocity.
If we are dealing with stereo, though, it does get more complicated, depending on how far you want to go with it. The head masks high frequencies, so real time filtering would be needed. Also it would be good to implement delay to match the different arrival times at each ear. And if you start talking about pinnas, I'm way out of my league.
As of now it seems like Pyroomacoustics does not support moving sound sources. However, do check a possible workaround suggested by the developers here in Issue #105 - where the idea of using a time-varying convolution on a dense microphone array is suggested.
I'm looking for an audio processing language or library which will allow me to experiment with different synthesis techniques. I've looked at Processing which I think is great at what it does, but haven't found any inspiring (and simple) audio libraries.
As a baseline, I want to simply create my own sample buffers and play them back (ideally in realtime). As a plus, the ability to handle MIDI events would be great. I'm an experienced C++ programmer so I could do it natively on but had hoped there was a more DSL (domain specific language) approach.
I have access to Windows, Mac or Linux so not too bothered yet about platform. Other languages I can deal with are C#, Java & Python.
Thanks
James
Depending on how much you want to stay out of the low-level housekeeping details, you may want to look at CSound , or if you want to not actually write code, the patching-based system PureData is great to work with. As #Lou points out, ChucK is interesting (but was too buggy to use the last time I checked it out).
If you really do want to write code, look at the Synthesis Toolkit, a set of C++ classes for audio processing and synthesis.
For an app framework, I recommend JUCE, which has incredibly nice cross-platform handling of audio/midi IO and GUI elements.
Max MSP is an audio production tool that is highly expressive.
I guess you could say it's a high-level tool, and not a low-level programming language. My impression of it is that it's geared towards the technical musician or the artistic engineer, but anyway it kicks ass and you could go low-level with it if you want.
I've always been a big fan of SuperCollider. It's designed for Mac OS X but also works on Linux.
The language is mostly based on SmallTalk, and it's pretty easy to pick up if you understand the basics of functional programming. The quality of the sound output by the SC Server is very good and there is plenty of documentation both built into the app environment and available online.
One interesting point of SuperCollider is the usage on android devices, and it's intercommunication with python trough out other modules.
Here goes an example
I know you didn't say Ruby, but check out Archaeopteryx
https://github.com/gilesbowkett/archaeopteryx/wiki
or ChucK
http://chuck.cs.princeton.edu/
Have a look at NAudio, an open source .NET audio SDK for working with audio files and devices in Windows. Some features include:
http://naudio.codeplex.com/
NAudio Features:
Play back audio using a variety of APIs
Decompress audio from different Wave Formats
Record audio using WaveIn, WASAPI or ASIO
Read and Write standard .WAV files
Mix and manipulate audio streams using a 32 bit floating mixing engine
Extensive support for reading and writing MIDI files
Full MIDI event model
Basic support for Windows Mixer APIs
A collection of useful Windows Forms Controls
Some basic audio effects, including a compressor
I have come up with an idea for an audio project and it looks like Go is a useful language for implementing it. However, it requires the ability to apply filters to incoming audio, and Go doesn't appear to have any sort of audio processing package. I can use cgo to call C code, but every signal processing library I find uses C++ classes which cgo cannot handle. It looks like libsox may work. Are there any others?
What libsox can provide and what I need is to take an incoming audio stream and divide it into frequency bands. If I can do this while only reading the file once, then bonus! I am not sure if libsox can do this.
If you want to use a C++ library you could try SWIG, but you'll have to get it out of Subversion. The next release (2.0.1) will be the first released version to support Go. In my experience the Go support is still a little rough, but then again the library I tried to wrap is a monster.
Alternatively, you could still create your own bindings through cgo using the same method SWIG does, but it will be painful and tedious. The basic idea is that you first create a C wrapper, then let cgo create a Go wrapper around your C wrapper.
I don't know anything about signal processing or libsox, though. Sorry.
There is a relatively new project called ZikiChombo
which contains so far some basic DSP functionality geared toward audio, see here
The dsp part of the project has filters on its roadmap, but they are not yet there. On the other hand some infrastructure for implementing filters, such as real fft and block convolution is there. Meaning that if you want FIRs, and can compute the coefficients by some other means, you can run them via convolution in zc currently with sound in real time.
Basic filtering design support (FIR,Biquad), for example using an ideal filter as a starting point will be the next step for zc. There are numerous small self-contained open source projects for basic and more advanced FIR and IIR filter design, most notably Iowa Hills which might be more accessible than a larger project to compute filter coefficients outside of Go.
More advanced filtering such as Butterworth, and filters based on polynomial solving and the bilinear transform will take more time for zc.
There is also some software defined radio Golang projects with some code related to filtering, sorry don't have the links offhand but a search for the topic may lead you to them.
Finally, there is a gonum Fourier package which also supplies fft.
So Go is growing some interesting and potentially stuff in this domain, but still has quite a ways to go compared to older projects (which are mostly in C/C++, or perhaps with a Python wrapper via numpy for example).
I am using this pure golang repo to perform Fourier Transforms with good effect
https://github.com/mjibson/go-dsp
just supply the FFT call with a
import (
"github.com/mjibson/go-dsp/fft" // https://github.com/mjibson/go-dsp
)
var audio_wave []float64
// ... now populate audio_wave with your audio PCM samples
var complex_fft []complex128
// input time domain ... output frequency domain of equally spaced freq bins
complex_fft = fft.FFTReal(audio_wave)
I'm wondering what is the recommended audio library to use?
I'm attempting to make a small program that will aid in tuning instruments. (Piano, Guitar, etc.). I've read about ALSA & Marsyas audio libraries.
I'm thinking the idea is to sample data from microphone, do analysis on chunks of 5-10ms (from what I've read). Then perform a FFT to figure out which frequency contains the largest peak.
This guide should help. Don't use ALSA for your application. Use a higher level API. If you decide you'd like to use JACK, http://jackaudio.org/applications has three instrument tuners you can use as example code.
Marsyas would be a great choice for doing this, it's built for exactly this kind of task.
For tuning an instrument, what you need to do is to have an algorithm that estimates the fundamental
frequency (F0) of a sound. There are a number of algorithms to do this, one of the newest and best
is the YIN algorithm, which was developed by Alain de Cheveigne. I recently added the YIN algorithm
to Marsyas, and using it is dead simple.
Here's the basic code that you would use in Marsyas:
MarSystemManager mng;
// A series to contain everything
MarSystem* net = mng.create("Series", "series");
// Process the data from the SoundFileSource with AubioYin
net->addMarSystem(mng.create("SoundFileSource", "src"));
net->addMarSystem(mng.create("ShiftInput", "si"));
net->addMarSystem(mng.create("AubioYin", "yin"));
net->updctrl("SoundFileSource/src/mrs_string/filename",inAudioFileName);
while (net->getctrl("SoundFileSource/src/mrs_bool/notEmpty")->to<mrs_bool>()) {
net->tick();
realvec r = net->getctrl("mrs_realvec/processedData")->to<mrs_realvec>();
cout << r(0,0) << endl;
}
This code first creates a Series object that we will add components to. In a Series, each of the components
receives the output of the previous MarSystem in serial. We then add a SoundFileSource, which you can feed
in a .wav or .mp3 file into. We then add the ShiftInput object which outputs overlapping chunks of audio, which
are then fed into the AubioYin object, which estimates the fundamental frequency of that chunk of audio.
We then tell the SoundFileSource that we want to read the file inAudioFileName.
The while statement then loops until the SoundFileSource runs out of data. Inside the while
loop, we take the data that the network has processed and output the (0,0) element, which is the
fundamental frequency estimate.
This is even easier when you use the Python bindings for Marsyas.
http://clam-project.org/
CLAM is a full-fledged software framework for research and application development in the Audio and Music Domain. It offers a conceptual model as well as tools for the analysis, synthesis and processing of audio signals.
They have a great API, nice GUI and a few finished apps where you can see everything.
ALSA is sort of the default standard for linux now by virtue of the kernel drivers being included in the kernel and OSS being depreciated. However there are alternatives to ALSA userspace, like jack, which seems to be aimed at low-latency professional type applications. It's API seems to have a nicer API, although I've not used it, my brief exposure to the ALSA API would make me think that almost anything would be better.
Audacity includes a frequency plot feature and has built-in FFT filters.