I am starting a project which would allow me to use Java to read sound samples, and depending on the properties of each sample (I'm thinking focusing on decibels at the moment for the sake of simplification, or finding some way to compute the overall 'volume' of a specific sample or set of samples), return a value from 0-255 where 0 would be silence and 255 would be the highest sound pressure (Compared to a reference point, I suppose? I have no idea how to word this). I want to then have these values returned as bytes and sent to an Arduino in order to control the intensity of LED's using PWM, and visually 'see' the music.
I am not any sort of audio file format expert, and have no particular understanding of how the data is stored in a music file. As such, I am having trouble finding out how to read a sample and find a way to represent its overall volume level as a byte. I have looked through the javax.sound.sampled package and it is all very confusing to me. Any insight as to how I could accomplish this would be greatly appreciated.
As Bastyen (+1 from me) indicates, calculating decibels is actually NOT simple, but requires looking at a large number of samples. However, since sound samples run MUCH more frequently than visual frames in an animation, making an aggregate measure works out rather neatly.
A nice visual animation rate, for example, updates 60 times per second, and the most common sampling rate for sound is 44100 times per second. So, 735 samples (44100 / 60 = 735) might end up being a good choice for interfacing with a visualizer.
By the way, of all the official Java tutorials I've read (I am a big fan), I have found the ones that accompany the javax.sound.sampled to be the most difficult. http://docs.oracle.com/javase/tutorial/sound/TOC.html
But they are still worth reading. If I were in charge of a rewrite, there would be many more code examples. Some of the best code examples are in several sections deep, e.g., the "Using Files and Format Converters" discussion.
If you don't wish to compute the RMS, a hack would be to store the local high and/or low value for the given number of samples. Relating these numbers to decibels would be dubious, but MAYBE could be useful after giving it a mapping of your choice to the visualizer. Part of the problem is that values for a single point on given wave can range wildly. The local high might be more due to the phase of the constituent harmonics happening to line up than about the energy or volume.
Your PCM top and bottom values would probably NOT be 0 and 256, more likely -128 to 127 for 8-bit encoding. More common still is 16-bit encoding (-32768 to 32767). But you will get the hang of this if you follow Bastyen's links. To make your code independent of the bit-encoding, you would likely normalize the data (convert to floats between -1 and 1) before doing any other calculations.