Or even better, how to get the size of the amplitude or the volume of the wave sound every certain time.
In fact I need the two ways, the full waveform and measure it each time. the first one for have a view of the song wave and the second one for visual effects.
this is for Android (NDK) systems.
come on people, I don't ask for the full code answer, I just want you to tell me some advices or something that can help me. You can simply say that the question is hard or makes no sense. but say something.
Whatever, I researched a little bit and I didn't find the answer for the question, but I did find a better solution for the problem, and is a free library named "superpowered", simple, fast, cross-platform, and has all the functions for analize sounds.
hope this help people new to this world of sound programming
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.
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.
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 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've tried looking up how I might go about this for a while now, and maybe I am using the wrong terminology in my searches or it's way too advanced for me. I basically want to be able to analyze audio files in real-time. I know hardly anything about audio processing so I should probably start small and work my way up. Eventually I'd like to be able to display a power (or frequency?) spectrum correlating to audio playing in real time. Basically like the WinAmp spectogram (terminology?)
Any online tutorials with perhaps an API suggestion or two would be greatly appreciated. I've found some vague explanations (mostly dealing with calculating FFT's then converting them to something...) Like I said, I know little of audio processing, so knowing where to start would be great.
Language of choice: C++
You could look into VST plugins as a starting point for the theory behind audio processing. There's a blog with some tutorials in c++ here.
You can also check out other SO questions on VST plugins for more info.
I believe audacity can run VST plugins, I'll look at that.
EDIT: Audacity doesn't support them out of the box, but you can enable it. You could download a trial of something like ableton live too.
I'd recommend using a graphical tool to begin with to prototype some ideas. Try Puredata or something similar.
http://puredata.info/
Juce is a fantastic way to get to grips with C++ with an Audio slant.
http://www.rawmaterialsoftware.com/juce.php
I've also stumbled across UGen which might help you get up and running without having to understand too much of the sample-by-sample processing theory. I've not looked at this much yet but it looks interesting at the outset.
http://code.google.com/p/ugen/
The KVR forums are full of knowledgable people who will help and direct newcomers to audio and plugin development.
http://www.kvraudio.com/
If you're feeling brave the dive in to a good book. I've heard a lot of good things about the following:
http://www.amazon.com/DAFX-Digital-Udo-246-lzer/dp/0471490784
Good luck! This is not an easy area to get going in!
(PS, the blog linked in the above answer is mine -> it's out of date and wont help you actually do any signal processing)