Free modulate software for linux

F Modular Synthesizer (FMS) is basically a tool to generate all kinds of sounds. It should work on any up-to-date Linux system.
Main features:
Playing sounds
- with any frequency and volume
- one after another or at the same time (mixer) or both
- stereo (different sounds on different channels)
- with any balance between the two channels
- with built-in sweep that even follows frequency modulation
- like speech synthesis [listen]
- both on-the-fly playing with direct user access and asynchronous, pre-computed output; also combining the two, taking the best of both
- with all modulation options described below
Making noise
- now although it tends to get noisy in any way, we mean real noise here
- in 4 different ways
- nearly-white noise
- narrow band noise
Modulating
- amplitude (volume)
- frequency
- minima, maxima, amplitude and frequency of the modulation (here the "Fully Modular" comes to existence)
- balance, moving the sound from left to right and back in stereo mode
- narrowness of the noise frequency band in many different ways
Making music
- some-kind-of-midi-mapping mode (sound + envelope => instrument, FMS midi file format)
- auto-converter and player for MUS file format, listen [here] (Musplay)
- auto-composer for those who like a rather modern kind of music
- chords
Displaying
- spectrum of a sound
- oscillogram of a sound
- mixes between different sounds with different volumes / frequencies
Saving
- sounds as oscillograms in FMS file format (mathematical description, not complete wavetables)
- sounds as Fourier overtone amplitudes
- music in FMS midi format
- FMS output in wave format
- wave file sounds in FMS file format (auto-converter) - experimental
GUI
- a somewhat ugly and extremely limited tcl/tk gui
- FROCOR, an installation that connects the FMS sound backend to graphical interfaces and outputs
- a fully modular Qt GUI - experimental, but Ill give you a [screenshot]
Enhancements:
- This release fixes all bugs discovered via user feedback, and should be quite stable.
- Sweep frequency modulation is now implemented, and using both blur and frequency modulation on the same sound at the same time is possible.

active acoustic vibration control tool is a set of basic tools for monitoring and modulating audio noise.

The tools aim toward a set of codes and algorithms for monitoring noise signal and modulation and reducing them using active noise control priciples.

Active noise control and soundproofing is a relativly recent development (first researched around 30 years ago), and has been in constant development for many and varied situations and applications.

The basic concept of active noise control is the aplication of equal and opposite force to cancel out vibration.

The project attempts to provide a starting point and library for experimentation into various methods of digital signal processing for active noise reduction and soundproofing.

Noise cancellation has been used in many environments currently active control is best used in constant vibrations with lower frequencys.

The principle is of controlling the noise level by adding an equal and opposite sound to the noise in the environment. The basic principle may be simple but the actual effectivness depends on many environmental factors.

Generally simplistic out of phase sound wave is used which works to some extent and requires little processing (a simple XOR with -1) this will tend to leave a large number of artifacts as the amplitude of higher frequency components (above i->o delay time) rises.

i->o processing delay
e transducer efficiency
p sound propagation characteristics
r reflections

The project aims to provide a framework for many dsp architectures, dac/adc chips, transducers and monitors.

Create tools for acoustic calibration and data logging hardware
To simplify implementation of active noise and vibration control systems.
To provide an open source standard for use within the industry and education.
To encourage hardware producers to distribute standard interface descriptions for their hardware
Enable low cost development of systems using COTS consumer hardware for noise and vibration control.

The Analysis & Reconstruction Sound Engine also known as ARSE, is a program that analyses a sound file into a spectrogram and is able to synthetise this spectrogram, or any other user-created image, back into a sound.
The ARSE consists in two main parts, a spectrographer with a base-2 logarithmic frequency scale, and a spectrogram synthetiser.
Unlike most spectrographers which are based on STFTs and perform the analysis by cutting the signal into small time slices to analyse these slices in the frequency domain, the ARSE is based on a filter bank followed by envelope detection, which means that the signal is cut into small frequency-domain slices, and then analysed in the time domain.
The filter bank is, as of now, made up with overlapping bandpass FIR filters defined logarithmically. Once the original signal is filtered with the filter bank, each resulting signal is sent to envelope detection.
Envelope detection in the ARSE isnt based on a Hilbert transform and peak detection, as its usually done. To achieve envelope detection, we first perform a FFT on the signal, zero-pad the beginning of the signal in the frequency domain according to a user-defined setting, then we perform an IFFT, and, now in the time domain, we turn every negative sample into a positive one, and we low-pass filter (and eventually decimate) the signal according to the same user-defined setting as we previously used.
For instance, lets say we have a signal with a sampling frequency of 44,100 Hz, and that we want to obtain an envelope for it which sampling frequency would be 100 Hz. Once we perform the FFT, we add enough zeroes in the frequency domain at the beginning of our signal so that every frequency component shifts by 50 Hz (100 Hz divided by two, it will later appear obvious why), and we perform an IFFT. Our signal now has a sampling frequency of 44,200 Hz (44,100 + 100 Hz), and the original signal which previously spanned from 0 Hz to 22,050 Hz now spans from 50 Hz to 22,100 Hz.
Now we turn every time-domain sample into its absolute value by turning every negative sample into a positive one. To perform this on a signal means that, for example, a sine wave of a certain frequency would become a signal which periodicity would be twice that frequency. Once we low-pass filter that signal to twice that frequency we obtain that signals envelope. In our case, now that we have obtained the absolute values for our signal, since the periodicity of a sine at the lowest frequency - 50 Hz - would now be 100 Hz, we only low-pass filter our signal at 100 Hz to obtain the original signals envelope. We can now decimate the signal to a sample rate of 100 Hz.
The resulting envelope for each frequency band makes the horizontal lines of the image representing the spectrogram. The amplitude of the envelopes translate linearly into intensity in the image.
The spectrogram synthetiser is based on modulation using horizontal lines of the image as envelopes. Each horizontal line is upsampled to the sampling rate of the desired final signals sampling rate, and is then modulated with, depending on the synthetisation mode chosen by the user, sines matching to the central frequency each horizontal line represents, or noise filtered through the filter bank.
Enhancements:
- Replaced fixed phase sine generation with random phase sine generation
- Changed the PRNG
- Removed the unused code
- Removed every call of nearbyint() due to compatibility issues
- Included the necessary files in order to make using ./configure && make && make install