This extension under development will work WITHOUT the need of an expensive B-format microphone, as the well known Soundfield microphone or the new AGM-MR2 microphone. One or several omnidirectional microphones is all that is required, and the B-format response will be computed from a set of independent Impulse Response measurements taken in a point and in several closely spaced positions. A temptative Fortran version (Character mode) of this forthcoming plugin can be freely downloaded here. It is called Crux7toBform.exe, and requires a 7-channels input. The final version will be a VST multichannel plugin, and will work with any number of microphones (4 to 24), producing 1st, 2nd or 3rd order B-format components (4, 9 or 16 channels).
This module will create the decoding impulse response matrix for ANY number of loudspeakers in ANY configuration. A typical Ambisonics listening setup makes use of 8 loudspeakers placed roughly at the vertices of a cube. A 4-channels, B-format signal has to be decoded in 8 feeds for the speakers. through a properly designed decoder. Although some analog and/or digital decoder is available on the market, their working principle is factory-preset to some typical speaker layouts, with a limited degree of freedom for adjustments on the field. The proposed system is completely self-calibrating: a B-format impulse response measurement is conducted for each loudspeaker of the reproduction rig. Then this set of B-Format IRs is processed through the software decoder, which produces the required set of decoding IRs (typically they are 32 for an 8-speakers system): these decoding IRs are usually very short (less than 1024 taps), making it possible to convolve them in realtime. Convolving the B-format original signal with this matrix of IRs, and mixing together the 4 streams for each speaker, the multichannel decoded wave file is produced. Then You have to play it through a multichannel sound card... . NOTE: the above process is capable not only of properly decoding an Ambisonics B-format signal, but it also perfectly equalizes and time-realignes the loudspeakers and is even capable of correcting for the room response (also if this is subject to the "dereverberation" constraints alread explained about the Cross Talk Cancellation module). This module will be released as a VST plugin - unfortunately, the current version of Audition only supports 2-channels VST plugins, so a true multichannel VST host (such as AudioMulch or Plogue Bidule) will be required. In the meanwhile, a small Fortran program which implements the same functionality, but computing just 1 or 2 speaker feeds at once, (VirtualMike.exe) is available for free download.
We have plan to develop a new import/export module, capable of reading and writing multi-channel files in the new Microsoft Wave-Ex format and in the Lossless-Compressed FLAC format. This will make it possible, for example, to employ directly (without any further file conversion) already existing freeware Ambisonics software decoders such as those produced by Richard Furse and by Richard Dobson, or to feed VST decoding plugins such as those developed by Gerzonic or Dave Malham. In the meanwhile, a simple Character-mode converter (Pack2x2to4.exe) which packs two stereo files in a single 4-channels file is freely downloadable from here. The final version will be available both as a FLT module for Audition and as a pair of multichannel VST plugins.
This module will replace the Audition's original Analyze/Frequency Analysis module, introducing more professional features, such as Magnitude & Phase display, separate left and right plots, calibration of the Y axis in dB-SPL or any other user-defined units, possibility to export the results onto the clipboard, choice of different analysis options (standard FFT, transfer function, power spectrum), capability to process only the selected waveform piece. A Beta version of this module is included in the Aurora 3.3 package - some of this functionality is also available in the Cross Function module of Aurora 4.0.
For the evaluation of tonal and impulsive components in environmental noise measurements, following the Italian regulations. This module will compute the 1/3 octave spectrum of the minima of the SPL-FAST values, and the temporal histories of the A-weighted SPL with the three time constants Slow, Fast and Impulse. The module will check these results against the Italian limits, and suggest the resulting penalties which have to be added to the measured Leq. NOTE: some of these functionalities are now included in the STI and ITUP56 modules!!!