I can't seem to find any information regarding the process that Ableton uses to efficiently detect atonal percussion and convert it into MIDI. I assume feature extraction and onset detection algorithms are executed, but I'm intrigued as to what algorithms. I am particularly interesting how its efficiency is maintained for a beatboxed input.
Cheers
Your guesses are as good as everyone else's - although they look plausible. The reality is that the way this feature is implemented in Ableton is a trade secret and likely to remain that way.
If I'm not mistaken Ableton licenses technology from https://www.zplane.de/ for these things.
I don't exactly know how the software assigns the different drum sounds, but the chapter in the live manual Convert Drums to New MIDI Track says that it can only detect kick, snare and hi-hat. An important thing is that they are identified by the transient Markers. For a good result you should manually check and adjust them. The transient Markers look like the warp Markers, but are grey.
compared to a kick and a snare for example, a beatboxed input is likely to have less difference between the individual sounds and therefore likely to be harder for Ableton to individually extract the seperate sounds (depends on the beatboxer). In any case, some combination of frequency and amplitude - more specifically(Attack, Decay, Sustain, Release) as well as perhaps the different overtone combinations that account for differences in timbre are going to be the characteristics that would have to be evaluated in order to separate the kick snare and hihat .
Before this feature existed I used gates and hi/low pass filters to accomplish a similar task. So perhaps Ableton's solution is not as complicated as we might imagine.
Related
I'm looking for a program that is able to recognize individual audio samples from my computer and reroute them to trigger WAV files from a library. In my project, it would need to be realtime as the latency would not be a desired result. I tried using dictation software that would recognize words to trigger opening a file and that's the direction where I want to go, but instead of words I want it to be sounds and it would happen in realtime. I'm not sure where to go and am just looking for some guidance. Does anyone have any suggestions of what I should do?
That's a fairly broad question, but I can tell you how I would do it. (Hardly the only way, but where I would start.)
If you're looking for real time input, the Java Sound library (excellent tutorial here) allows for that. (Just note that microphone input from a web page is difficult on anything, due to major security concerns, so this would be a desktop application.)
If it needs to be real time, the first thing I would suggest is stream and multithread the hell out of it. I would suggest the Java 8 Stream API, but since you're looking for subsamples that match a specific pattern, then each data point will have to be aware of the state of its neighbors, and that isn't easy with streams.
You will probably want to know if a sound roughly resembles an audio profile, so for that, I would pick a tolerance on just how close you want it to be for a match (remembering that samples may not line up 100% anyway, so "exact" is not an option), and then look up Hidden Markov Models. I suggest these because they're what voice recognition software typically uses, and while your sounds may not be voices, it will give you an idea of what has already been done.
You'll also want to maintain a limited list of audio samples in memory. Specifically, you will likely need the most recent data, because an audio signal is a time-variant signal, and you can't get a match from just one point. I wouldn't make it much longer than the longest sample you're looking to recognize, as audio takes up a boatload of memory.
Lastly (for audio), I would recommend picking a standard format for comparison. Make it as good as gets you decent results, and start high. You will want to convert everything to that format before you compare it.
Once you recognize a specific sound, it's basically a Command Pattern. Specific sounds can be mapped, even with a java.util.HashMap, to specific files, which (if there are few enough) you might even have pre-loaded.
Lastly, it's worth looking at the Java Speech API. It's not part of the JDK and it's quite dated, but you might get some good advice from its implementation.
This is of course the advice of a Java-preferring programmer, but I imagine that there might be some decent libraries in Python and Ruby to help you as well; and of course there's something in C somewhere. This may sound like a lot, but most of the material is already implemented and ready-to-go.
Hopefully this helps, let's look forward to other answers.
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.
I am looking for some advice on categorizing a library of sound effects. I have a large set of random sound effects, (think whistles, pops, growls, creaks, gunshots etc). I would like to be able to take a growl for example, and find the next growl that sounds the closest to the original.
Given a sound, what sound from my set sounds the closest to it.
I have done a fair amount of googling and have found two avenues that I am still researching. One is using echonest, although their "best match" support looks not promising for public users. The other option is diving into FFT and building my own matching algorithm. This is a fine option and would be a great learning experience but I wanted to get some opinions from others who might know a little more about sound processing; especially short clips .5sec - 3sec range, not full length music.
Thanks!
I have worked in movie postproduction for years and as far as I know, there is no way to do that automatically. Every file has meta information in its file header which describes what the sound is like. You are then actually not searching for the file names but in the meta string.
I don't think that it is trivial to sort effects programmatically as two effects that sound similar might be totally different if you look at the waveform.
You would need to extract significant information about a sound that you can then compare.
I am also not a DSP expert, maybe there are methods to do this
If you're interested in trying to build your own system to do this, I can suggest a few keywords that might help to refine your Google searches. In the academic research community, the task you're describing is often called "content-based audio searching". I know there's been a lot of work done on it, and though most pertains to music, sound effects have definitely been the focus of a number of studies.
You might want to start with the work of Pedro Cano.
Also, I recently heard about a company that's doing similar work. You might want to check out products from Imagine Research.
Those are just a couple of ideas off the top of my head. I'm not %100 sure they'll be helpful. If they are, please let me know!
I'd like to be able to algorithmically create sounds (like monster growls, or distant thunder.) This isn't as widely covered on the net like more traditional procedural content (terrains, etc.) Any one have any algorithms on how to create these kinds of sounds?
This, in general, is a very hard problem. Just like drawing, each sound is its own thing, and needs its own algorithms, and, like drawing, some are more easily done by algorithm than others. There's no general algorithm for creating sound any more than there's a general algorithm for drawing all things like faces, insects, and mountains. Each is it's own project (and often quite a big one), unless you're just looking to draw circles or generate sine waves.
Most of the case studies I know of are the many attempts to generate musical instrument sounds, and generally each of these attempts is a PhD thesis.
For a time-efficient solution, sampling is the way to go.
Or, if you really need a procedural approach, you could ask the question for one specific type of sound, and people might be able to come up with an algorithm for it. For example, I'd be interested in taking a shot at a "distant thunder" algorithm, but don't want to bother if having just thunder but no monsters, etc, is not useful to you.
I would suggest checking out the many software projects and papers of Perry Cook who has done some great work in the realm of physical modelling (though his website is a bit of a nightmare to navigate). Though as tom10 says, it's a very hard area. If you have the stomach for a bit of signal processing then it's a very fascinating area to get into.
My question is not completely programming-related, but nevertheless I think SO is the right place to ask.
In my program I generate some audio data and save the track to a WAV file. Everything works fine with one sound generator. But now I want to add more generators and mix the generated audio data into one file. Unfortunately it is more complicated than it seems at first sight.
Moreover I didn't find much useful information on how to mix a set of audio samples.
So is there anyone who can give me advice?
edit:
I'm programming in C++. But it doesn't matter, since I was interested in the theory behind mixing two audio tracks. The problem I have is that I cannot just sum up the samples, because this often produces distorted sound.
I assume your problem is that for every audio source you're adding in, you're having to lower the levels.
If the app gives control to a user, just let them control the levels directly. Hotness is their responsibility, not yours. This is "summing."
If the mixing is automated, you're about to go on a journey. You'll probably need compression, if not limiting. (Limiting is an extreme version of compression.)
Note that anything you do to the audio (including compression and limiting) is a form of distortion, so you WILL have coloration of the audio. Your choice of compression and limiting algorithms will affect the sound.
Since you're not generating the audio in real time, you have the possibility of doing "brick wall" limiting. That's because you have foreknowledge of the levels. Realtime limiting is more limited because you can't know what's coming up--you have to be reactive.
Is this music, sound effects, voices, what?
Programmers here deal with this all the time.
Mixing audio samples means adding them together, that's all. Typically you do add them into a larger data type so that you can detect overflow and clamp the values before casting back into your destination buffer. If you know beforehand that you will have overflow then you can scale their amplitudes prior to addition - simply multiply by a floating point value between 0 and 1, again keeping in mind the issue of precision, perhaps converting to a larger data type first.
If you have a specific problem that is not addressed by this, feel free to update your original question.
dirty mix of two samples
mix = (a + b) - a * b * sign(a + b)
You never said what programming language and platform, however for now I'll assume Windows using C#.
http://www.codeplex.com/naudio
Great open source library that really covers off lots of the stuff you'd encounter during most audio operations.