BrahmaVolver

Contents

• 1. Introduction
• 2. How I can use it
• 3. How it looks to me
• 4. How it works
• 4.1 The filtering matrix
• 4.2 The 5.1 conversion
• 5. FAQ

This is a piece of software that converts an A-format four channel recording taken with a Brahma recorder in

• B-format, useful for 1st-order Ambisonics systems
• 5.1 (5.0) format

1. Select the input mode on the left; if your four channel recording belongs from a Brahma digital recorder, your tracks should have been stored in '2 stereo file'
2. Set files to convert; you can type directly the file(s) name(s) (press Enter after each name!), use the file browser button, or drag files on the file name boxes
3. Select the output type on the right: B-format or 5.1
4. Optionally you can setup the output mode pressing the left arrow button; the setup window will opens and shows you the possibility to export the converted tracks in three ways: one single multichannel wav file, two/three stereo files and four/six single channel (mono) files. In the 5.1 case, you can set also
   • the aperture angle of the front left/right speakers
   • the use of the effect channel as low frequencies channel.
   The setup window can be closed with the left arrow button.
5. Press Convolve button.

Input stuffs are an the left, and output ones are on the right; every item has its own tooltip so you can easily see what it does.

In the bottom a status bar informs you about:

1. the current state of the program
2. the working sample frequency (in Hz)
3. the input file length (in samples)

From the menu File you can just quit the program; slightly more things in the menu Settings: you can set the Full Autorange options for the filtering routine and Reset all the input/output fields.

BrahmaVolver is a four channel filtering tool: this means that it takes as input a vector which elements are the four recorded tracks and gives out another four elements vector which elements are the processed tracks.

Using some math, if x is the input vector, and h the 4 x 4 filtering matrix, the processed output y is given by:

The big issue is the computational effort requested by a convolution operation, for this reason, filtering is usually implemented in the frequency domain, where convolution become a simple product. If X, H and Y are the Fourier transforms of, respectively, input vector, filtering matrix and output vector, we have:

The filtering matrixes that come with the software are suitable for an A to B format conversion, but if you want to do some others elaborations, you can substitute it.

When Convolve button is pressed, BrahmaVolver searches for the file 'FilterMatrix_xxxxx.wav' in the filters subdirectory, where xxxxx is the sampling frequency in Hz detected from input files; if the filter file doesn't exist, no calculation can have place.

The format of the filtering matrix is simply a four track WAV file where the tracks are the rows of the matrix and the columns are appended one after the other

Important: each item must have the same length!

to be written

Q: I get the message 'The number of channels doesn't match'.

A: This means that one of your input files (or both) is not stereo. BrahmaVolver can manage only files recorded with a Brahma digital recorder that produces four channel (A-format) recordings splitted in two stereo files.

Q: I get 'Selected file has different sampling frequency than others already stored'

A: Input files must have the same sampling frequency; be also sure that you have a filter with the sampling frequency of your input files. I.e. if your files are sampled at 48000 Hz, in filters subdirectory must be a file named 'FilterMatrix_48000.wav'.

Q: This is annoying: 'Selected file has different length than others already stored'

A: Yes, but quite obvious: if your input recording is splitted in two or four files, these should have the same length (have you recorded them in the same moment or not?), if this is not true, your files can belong from different recordings...