![]() A new method is then demonstrated using the Tabu search algorithm coupled with lateralisation parameters extracted from a binaural simulation of the Ambisonic system to be optimised (as these are the parameters that the Vienna system is approximating). ![]() During the write up of this report Craven (2003) has shown how 4th order circular harmonics (as used in Ambisonics) can be used to create a frequency independent panning law for the five speaker ITU array, and this report also shows how the Tabu search algorithm can be used to optimise these decoders further. A method, based on the Tabu search algorithm, is applied to the Vienna decoder problem and is shown to provide superior results to those shown by Gerzon and Barton (1992) and is capable of producing multiple solutions to the Vienna decoder problem. In this work, the original work by Gerzon and Barton (1992) is analysed, and shown to be suboptimal, showing a high/low frequency decoder mismatch due to the method of solving the set of non-linear simultaneous equations. Although Ambisonics has been well researched, no one has, as yet, produced a psychoacoustically optimised decoder for the standard irregular five speaker array as specified by the ITU as the original theory, as proposed by Gerzon and Barton (1992) was produced (known as a Vienna decoder), and example solutions given, before the standard had been decided on. The conversion from a five speaker ITU array to binaural decode is well documented but pair-wise panning algorithms will not produce the correct lateralisation parameters at the ears of a centrally seated listener. ![]() In order for this system to function optimally, each of the three systems rely on providing the listener with the relevant psychoacoustic cues. The need for a system which is platform independent is discussed, and the proposal for a system based on an amalgamation of Ambisonics, binaural and transaural reproduction schemes is given. A literature review of psychoacoustics and a review (both historical and current) of surround sound systems is carried out. This thesis describes a system that can be used for the decoding of a three dimensional audio recording over headphones or two, or more, speakers. This was justified by objective analysis, where the convolution reverb shown to be having a faster decay rate for high frequency bands, thus sounding unnatural. However personal preference category resulted in 61%of the subjects preferring the chamber reverb. Null hypothesis was proved, where the realism difference of the two samples resulted in 52% of the participants preferring chamber reverb (control signal). Listening tests were designed so that realism, quality and personal preference categories were present. The samples were also analysed in their spectrogram views in order to analyse the quality objectively. These samples were then used for subjective analysis in a pair-wise categorical preference type listening test. Also, using same equipment and setup, an impulse response was captured in the same chamber to form the test sample. In order to realise, anechoic drum kit music samples were recorded and re-recorded in a chamber to form the control signal. An experimental design methodology was introduced in order to evaluate the perceptual quality of convolution reverb. The real-time fast convolution reverb plugin was implemented using multiple frequency delay line non-uniform partitioned convolution method on MaxMSP. This project is based on creating a convolution reverb plugin on MaxMSP and evaluating the perceptual quality of convolution reverb. The product of this operation is a third signal, containing reverberation of the space where the impulse response is captured. Convolution reverb is the process used for reverberating a signal by an impulse response of an actual space.
0 Comments
Leave a Reply. |
Details
AuthorWrite something about yourself. No need to be fancy, just an overview. ArchivesCategories |