These are command line programs for manipulating audio. They expect input on stdin and produce output on stdout. They all deal with raw 32-bit float audio. Floating-point samples are nice because they can go above 1.0 and not clip.
Eventually, the goal is/was to replace all of this with the more modular Truesynth project, but I continue to use these programs, so why not release them to the world.
unfmt -16 <mysong.raw | amp -dB +6 | softsat | fmt -16 >louder.raw
This assumes mysong.raw is a raw waveform with 16-bit integer samples, like a typical .wav with the header stripped off.
unfmt -16 takes 16-bit integer samples and converts them to 32-bit
float. It's worth knowing that while 16-bit integer samples (by
definition) range from -32768 to 32767, these utilities all treat -1.0
to 1.0 as full-scale. So unfmt -16 is not only converting, but also
dividing by 32768.0.
amp -dB +6 increases the level by six decibels, which is about double
the amplitude (you could also use amp -vol 2.0, but usually dBs are
more friendly). Fun fact: While you can double an amplitude, there's not
really such a thing as "twice as loud" — the ear doesn't work that way.
softsat is a soft clipper. This effect lowers all samples into the
acceptable -1.0 to 1.0 range while adding less distortion than you would
get from hard clipping.
fmt -16 does the opposite of the unfmt command.
You can build each program with the provided build script. To build
them all, use it like this:
$ mkdir bin
$ cd bin
$ ../build ../*.c
This cleanly puts all binaries in the bin/ directory. You can also
build a single one with ./build programname.
I have never tested this software on a big-endian system. fmt and
unfmt are the only programs that will care (since all the others use
floats), and you may have to change them to do what you want.
To find out what options a program takes, run it with the -help
option. All programs assume 44100 Hz sample rate by default, which you
can override by setting the RATE or SR environment variable; for
example, type export RATE=48000 in the shell before running these
programs.
- envelope: (with -generate option or if stdin is a tty)
- octagon: octagon wave (between square & tri) oscillator
- rossler: rossler oscillator
- saw: up/down saw wave oscillator
- sine: sine wave oscillator
- square: square wave oscillator
- tri: triangle wave oscillator
- white: white noise
- stretch: time and pitch stretcher, or resampler
I give this its own category because it is the only effect where number of samples in and out won't be equal.
- amp: gainer
- clip: hard clipper
- comp: peak/RMS compressor
- delay: delay line
- delay2: delay line with LFO
- deriv: differentiates input
- envelope: applies an envelope to input (can also be a generator with the -generate option)
- fadef: add fade-ins or fade-outs
- filter: IIR biquad filter (highpass, lowpass, bandpass & more)
- foldback: foldback distortion
- gate: a simple "gate"
- int: integrates input
- limit: crappy limiter, use limit2 instead
- limit2: a fairly decent lookahead limiter
- midside: left/right -> mid/side stereo converter
- pan: panner
- pink: FIR filter designed to convert white to pink noise
- power: raises all samples to a power (i.e., a distortion effect)
- reverb: a brute-force, unsophisticated but useful reverb
- sampbins: randomized sample-and-hold effect
- sfclip2: soft-clipper (avoid, softsat is better)
- shape: waveshaper distortion
- sine: with -modfreq option, input modulates sine frequency
- sineclip: weird noise effect, clips to shape of sine wave
- sinshape: strange sine-based waveshaper
- softclip: soft clipper (avoid, softsat is better)
- softsat: soft saturator / limiter
- stwidth: stereo width adjuster
- mix: sum waves
- mul: multiply waves
- pmix: sum the output of commands
- fix: set float samples that are NaN (not-a-number) or infinity to zero
- fmt: convert floats to integer samples
- unfmt: convert integer samples to floats
Released under CC0 Public Domain Dedication. See the LICENSE file for details.