38 lines
1.6 KiB
Markdown
38 lines
1.6 KiB
Markdown
# Library for Acoustic Signal Processing
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Welcome to LASP: Library for Acoustic Signal Processing. LASP is a C
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library - currently still under heavy development - with a Python
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interface which is supposed to process (multi-) microphone acoustic
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data in real time and output results.
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The main goal of this library will be the processing of data from an
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array of microphones real time, on a Raspberry PI. At the point in
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time of this writing, we are yet unsure whether the Raspberry PI will
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have enough computational power to this end, but may be by the time it
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is finished, we have a new faster generation :).
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Current features that are implemented:
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- Compile-time determination of the floating-point accuracy (32/64 bit)
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- Fast convolution FIR filter implementation
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- Decimation of the sample rate by an integer factor of 4.
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- Octave filterbank FIR filters designed to comply with IEC 61260
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(1995).
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- Averaged power spectra and power spectral density determination
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using Welch' method. Taper functions of Hann, Hamming, Bartlett and
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Blackman are provided.
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- A thread-safe job queue including routines to create worker threads.
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- Several linear algebra routines (wrappers around BLAS and LAPACK).
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- A nice debug tracer implementation
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Some of the near future features:
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- Third octave filter bank
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- Slow and fast time updates of (A/C/Z) weighted sound pressure levels
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- Conventional and delay-and-sum beamforming algorithms
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For now, the source code is well-documented but it requires some
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additional documentation (the math behind it). This will be published
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in a sister repository in a later stage.
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If you have any question, please feel free to contact us: info@ascee.nl.
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