The http://dreamscape.68.org/~michael/audiofile was originally proprietary to SGI, and was used in Irix to read and write audio files of various types: wav, aiff, au, etc. After a bit of coding by Michael Pruett, and a few more things, it's no longer proprietary, though some parts are not yet in existence. We'll talk about that later. But here's
This extension is a binding of the audiofile library to the Ruby language. With the aid of this super-wonderful extension, you can do everything you could do with audiofile in Ruby, except:
The last item may need a small bit of explanation: there are a number of functions declared in audiofile.h (or commented out) which show that the author has a clear idea of what it will look like to the user of the library to use these not-yet-supported features, but there's not actually any code behind these declarations.
Version 0.2 adds file writing support, and fixes a bug with #read where it returned half the data it was supposed to. Also the API has changed some; the real_ prefixes no longer exist and there is a new virtual_ prefix. This makes code using ruby-audiofile shorter and more succinct.
Version 0.2.1 takes out a couple of debugging printf's i accidentally left in, and changes the width/width= methods to bits/bits= to fit better with Linux::SoundDSP. Minor documentation changes as well.
Either inside a Ruby source tree, or inside the Ruby library directory (for me, /usr/lib/ruby/1.6/i386-linux), untar the archive, change into the audiofile directory, and run these commands:
ruby extconf.rb make su make install
In your scripts,
require 'audiofile'
This defines the AudioFile class, which has the following methods:
Open a new AudioFile and return it. Hopefully, the filename needs no explanation. The mode is either "r" or "w". The filesetup is only needed when writing files, and describes the sample-rate, sample-type, file type, etc.
If you're writing a file, you should know how writing actually happens. When you open a file for writing using the audiofile extension, it is not actually opened until you write to it. This is so that you can set the properties of the file (file format, sample width, number of channels, etc.; see below for these methods). If you do not set all of the properties of the file, you will get undefined results. The properties that must be set are (in order of the length of their names):
Close the file.
Read frames frames from the file. A frame is one sample for each channel. So in a 44100Hz, 16-bit, stereo file, a frame would take four bytes: two for the left channel, and two for the right. (Two bytes is 16 bits of course.) This returns a new string every time, so if you're reading many times from an audio file, read_into is suggested. That way your script will not grow huge in memory.
Read into string from the file. This replaces the contents of the string by reading the largest number of frames that will fit into the current length of the string. It returns the number of bytes actually read. Example use:
string = " " * 32 file.read_into(string)
Write the string to the file. See the note under #open for how this happens and how that affects you.
Also note that both #read_into and #write round to the frame. That is, read_into will only read as many whole frames as fit into the string you give it, and write will only write as many whole frames as are contained in the string you pass to it.
Flush the write buffers for the file.
Returns the current position inside the file, in frames. (See #read about frames.)
Move to a new position inside the file. The position is specified in frames. You cannot move backwards inside the file; #pos= will throw an exception if you do.
Returns how many frames are in this file.
Sets the virtual byte order. new_byte_order should be either AudioFile::BIG_ENDIAN or AudioFile::LITTLE_ENDIAN.
When you read from the file, this byte order is the one the results will come back in. So if you are reading a little-endian file and you use #byte_order= AudioFile::BIG_ENDIAN, then #read, the data you get will be in big-endian byte order, because the library will switch the bytes on the fly.
Conversely, if you are writing a file, and you set the virtual byte order, and write to the file, the library will use the bytes you give it in the virtual byte order, and swap them as needed in order to get them in the file in the real byte order.
Returns the virtual byte order. This will be one of:
Returns the real byte order. See the constants above. This is the byte order in which the file is stored and does not change when you use
Sets the real byte order. For use when writing files only. Use before actually writing to the file.
Returns the real compression type, which will be one of:
or one of these, all of which are unsupported by the audiofile library:
Sets the real compression type. See the constants above. For use when writing files only. Use before actually writing to the file.
Returns the real sample format, which will be one of:
Sets the real sample format. See the constants above. For use when writing files only. Use before actually writing to the file.
Returns the number of bits in a sample. Commonly, this is 8 or 16. Some of the file formats supported support sample widths greater than 16, which are usually 24 or 32. The default behavior of the library is to pad 24-bit samples to 32 bits for speed of handling. To change this, look for the EXPAND_3TO4 define in the source (audiofile.c, around line 90), change and recompile.
Sets the number of bits in a sample. For use when writing files only. Use before actually writing to the file.
Returns the number of samples per second. (commonly 44100, 22050, etc.)
Sets the number of samples per second. For use when writing files only. Use before actually writing to the file.
Returns the number of channels. (commonly 1 or 2)
Sets the number of channels. For use when writing files only. Use before actually writing to the file.
param is an array containing four floats, corresponding to the slope, intercept, minimum clip and maximum clip values. This alters how the library sees the samples. Example:
file.real_pcm_mapping = [0.0, 1.0, -1.0, 1.0]
You now know almost as much about this as I do. (Which isn't much.)
The audiofile library (ideally) supports complete virtualization of all parameters of a sound file (sampling frequency, sample type, compresson...) such that the library would translate on the fly from the format in the file to the format the user of the library wishes to read. All of this support isn't in place yet: in fact, the only thing that is virtualized right now is the byte order. So when complete virtualization is supported, my binding will have
there's only #virtual_byte_order and #virtual_byte_order=.
If you have any more questions about AudioFile, you can look at the source (by the way, many thanks to matz for writing extensions before I did; I copied the structure of my extension from gdbm, which he wrote). Or you can ask me questions at mailto:liver@soon.com. Or try to catch me on irc.openprojects.net channel #ruby-lang, as sarynx. (If it's not during the [Northern Hemisphere] summer you might not see me much there. Go for the email.)
If you use this extension in a Ruby app, I'd love to know! Email me and tell me.
Might support querying the library for its capabilities. Might support non-strictly-audio data parts like instruments and loops.
The methods that return constants will hopefully end up more elegant in some way.