a tool for analyzing audio signals in terms of its content. It is an opensource project released under
It aims to have the following capabilities.
Replicate human hearing
system in terms of separating sources, recognizing speech and thus
analyze the auditory scene.
Listen to the separated
It will accept inputs from
files or a two microphone array setup whose construction would be
described along with the project or else were.
These processes can be done
on any segment of the speech which has been input to the program.
Other than this specialized
function the software would do other processing like, synthesizing
spatialized sounds from mono recordings, saving, cutting, splicing,
normalizing, analysis via spectrogram, filtering using various kinds
of filters and re-sampling.
In its present stage of
development it has the capability to
Record and save files, mono
Normalize the the signals
Play the processed signal
Separate sources using recordings
from a two microphone array (It is not real time at this stage so
recording has to be done as a first step and the processed)
View and listen to the separated
sources and if necessary save them.
Program is supported on
Linux. It is
- written in C
- xforms from T.C. Zhao and Mark Overmars
- libsndfiles by Erik de Castro Lopo
- fftw3 libraries from MIT.