概述
Installing the application requires admin privileges.
Digital Signal Processing (DSP) software.
Spectral analysis (Fast Fourier Transform).
FIR filters: equiripple multiband linear-phase FIR filter (Remez / Parks–McClellan algorithm), Kaiser window,
IIR filters: second order poles and zeros, Butterworth low pass, high pass, band and notch. First order.
Artificial reverberation: feedforward and low pass feedback comb filters, Schroeder allpass sections.
IIR/FIR filters from coefficients stored in files. Both with optional low pass filter taps (first order IIR filters).
Export filter coefficients to files.
DSP filters from Linear Predictive Coding (LPC) coefficients with full pole decomposition: extract glottal waveform from speech record.
Filters can be applied in real-time while an audio track is playing.
Their poles and zero of any filter can be visualized in both Laplace (frequency) and Z-Transform domains.
Audio editor with audio track edition and sample rate conversion.
Multiple sine wave and white noise generator.
Loopback recording capability that allows you to capture the audio stream that is being played by another media player or a web browser.
Acoustic module dedicated to complex vocal tract modelling.
Acoustic of interconnected acoustic ducts of varying cross-section:
Compute the pole (resonance) frequencies (formants) and bandwidths.
Compute the transfer function between two points of the acoustic device.
Liljencrants-Fant glottal wave model for real-time articulatory speech synthesis.
Both a desktop application and a Python library module are available in the package.
The Python module allows you to access all the application features from your Python scripts
and integrate it into your Python API or application.
特性
Multiple sine wave and white noise generator
Spectral analysis: Fast Fourier Transform and Linear Predictive Coding.
Real-time digital filters: second order Pole and Zero, Butterworth, Kaiser Windows (FIR) First Order IIR, reverberation ( Comb and Allpass ).
Liljencrants-Fant Glottal Wave Model
Acoustic of interconnected ducts
Real-time articulatory speech synthesis
Linear Predictive Coding (LPC) with pole decomposition
Glottal waveform from speech record
Sample rate conversion
Python Bindings: record what you do in the Application execute it afterwards thanks to an embedded Python interpreter.
Loopback Recording: capture the audio stream that is being played by another media player or a web browser.
Reverberation: Comb and Allpass filters.
Equiripple linear-phase FIR filters (Parks-McClellan/ Remez algorithm).
Import/export DSP filters from coefficients stored in /to files.
此版本中的新增功能
Equiripple multiband linear-phase FIR filter (Remez / Parks–McClellan algorithm).
Import/export DSP filters coefficients from files.
系统要求
适用平台:电脑
操作系统:Windows 10 版本 0.0 或更高版本
键盘:集成键盘
鼠标:集成鼠标
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