This is a JUCE port of the mda audio plugins collection from Paul Kellett.
The original mda audio processing plug-ins in VST format were available for many years as closed-source freeware from mda-vst.coms.
The available online version are:
- the original repo on Sourceforge which includes an AudioUnit port by Sophia Poirier
- an old port to VST2.x SDK
- a newer port to VST3.x SDK
Therefore I decided to port this collection to the modern JUCE framework as it would be the best way to learn after the tutorials and documentation.
Below is a description of each plugin as it was originally provided.
This plug-in simulates a guitar amplifier. The distortion section is very basic and you may get better results by pre-processing the signal with EQ or another distortion effect, and using mda Combo for speaker simulation.
The Model parameter selects the speaker type:
- D.I. - direct from the distortion, for a fuzz sound
- Spkr Sim - simple low pass filter speaker simulator
- Radio - transistor radio speaker / telephone
- MB - single speaker with close or more distant mic
- 4x12" - large 4-speaker cabinet with front or side mic
The HPF parameter (not present on the original VST version) is a high-pass filter before the distortion stage, which can make the distortion much sweeter sounding.
The Drive parameter controls the amount of distortion, and the Density parameter adjusts the saturation curve between hard/clipped and soft/full. The Bias parameter adds a DC offset to the distortion stage, which can give the distortion a gated effect or add more even harmonics. Lastly, the output parameter allows the output level to be adjusted.
This plug-in reduces the bit-depth and sample rate of the input audio (using the Quantize and Sample Rate parameters) and has some other features for attaining the sound of vintage digital hardware. The headroom control is a peak clipper, and the Non-Linearity controls add some harmonic distortion to thicken the sound.
Post Filter is a low-pass filter to remove some of the grit introduced by the other controls.
This plug-in is a pitch shifter designed to produce the classic detune effect, where the left channel is shifted down in pitch and the right channel is shifted up. This sounds similar to a chorus effect, but with less obvious modulation.
The delay inherent in the pitch shifting process can be adjusted with the Latency control - longer settings are needed to make low frequency signals sound smoother, but can also adds a nice doubling effect to vocals.
A simple delay line with filtering and saturation in the feedback loop. When the feedback mode is set to "saturate", soft saturation is used to limit the level of the delay repeats. When set to "limit", hard clipping is used. The Feedback Tone parameter filters the feedback signal with a highpass or lowpass filter as it is turned to the right or left respectively. The LFO Depth and LFO Rate parameters introduce some pitch modulation, as does the Delay parameter when it is moved.
This plug-in is a very ordinary analog-style compressor and noise gate, designed to sound like the dynamics section from a large- format mixer. A peak limiter is preset to 0 dB FS (the compressor release time parameter also sets the limiter release).
This plug-in is a simulation of a Leslie rotary speaker. Actually it is more of a caricature, giving the impression of a rotary speaker rather than being a strict simulation. Stereo Width and "Throb" (amplitude modulation) can be adjusted for both the high and low frequency rotors. The traditional Slow/Fast/Stop speed control is provided with an additional speed parameter for fine tuning.
This unusual effect is a triggered pitch-shifter. It works by detecting transients, and between each transient recording the input signal while playing it back at a slower rate. This shifts the pitch downwards, but if the Threshold is set correctly the output can remain tightly synchronised with the original signal. The Tune and Fine parameters adjust the pitch shift amount. The Decay parameter makes the signal fade out after each trigger, which can be useful while setting the Threshold control. The Hold parameter sets the minimum time between triggers, and the Mix control allows the wet and dry signals to be blended, which can give an interesting "triggered flanger" effect when the mix is set to 50%. When the Quality parameter is set to "High", the pitch shifting quality is improved. This effect can be useful on drums or whole rhythm sections. It can also be used to add shuffle to a straight groove, by setting the Threshold and Hold parameters so it only retriggers on every other beat.
This was the first "mda" effect, made way back in 1998. It is a simple ring modulator, multiplying the input signal by a sine wave, the frequency of which is set using the Frequency and Fine Tune controls. As if ring modulation wasn't harsh enough already, the Feedback parameter can add even more edge to the sound!
This plug-in is a 3D Autopan which can give the impression of the sound circling the listener. The Rate control sets the autopan speed, and the Pan control sets the pan position when the rate is zero. Like all 3-dimensional effects, be sure to check the resulting sound for mono compatibility!
This plug-in generates a continuously rising or falling tone. Or rather, that's what it sounds like but really new harmonics are always appearing at one end of the spectrum and disappearing at the other. (using some EQ can improve the psychoacoustic effect, depending on your listening environment). These continuous tones are actually called "Risset tones", developed from the earlier "Shepard tones" which change in series of discrete steps. The Mode control allows the input signal to be mixed or ring modulated with the tone - this works well as one element of a complex chain of effects.
This plug-in is a high resolution vocoder, designed for a natural rather than an electronic sound. The Quality parameter is equivalent to the number of bands in a normal "channel" vocoder. The plug-in expects a modulator signal (usually voice or drums) in the main input bus and a carrier (usually synth or guitar) in the second input bus, but the Input Swap parameter can be used to reverse this.
*** Please note that Talkbox is a multi-input-bus plug-in and not all AU hosts support plug-ins with more than 1 input bus. Currently Logic, Live, and Digital Performer are known to support this. In Logic and Digital Performer, you can route the second input bus in via the plug-in's "sidechain" menu selection. In Live, you will see the second input bus appear as an audio output destination on tracks.
For a typical talkbox sound, try inputting a synth sound with oscillator sync (at a fixed slave frequency, between 12 and 20 semitones above the master) with the master oscillator volume turned down, and vibrato added with the mod wheel or aftertouch. Alternatively, for a natural vocal sound use a low-pass filtered saw wave with random pitch modulation, plus high-pass filtered white noise.
This plug-in produces a range of high-precision audio test signals. The Level parameter sets the peak level of the sine wave output in dB FS. As the noise signals do not have a well defined peak level they have been aligned to the same RMS level as the sine wave, i.e. 3.01 dB below the peak level. Warning: The noise signals can have peak levels more than 10 dB above their RMS level so will clip at high level settings.
Linear sweeps start at 0 Hz and end at the indicated frequency. If the indicated frequency is above 900 Hz, log sweeps start at 20 Hz and end at the indicated frequency, else they start at the indicated frequency and end at 20 kHz.
This plug-in analyses the frequency of the input signal, and uses that frequency to control an oscillator (with sine, saw or square waveform) a ring modulator (the input signal is ring modulated by a sine wave) or the centre frequency of a peaking EQ. The Dynamics control adjusts how closely the oscillator output level follows the input signal level. The Glide control smooths the pitch tracking, and the Transpose control adds a pitch offset to create musical intervals with the input signal. The Maximum control limits the maximum pitch, which can help reduce glitches in the pitch tracking, and the Trigger contol sets a threshold below which the pitch stays constant, which can help reduce glitches and unwanted gargling when the input signal is quiet. A new control for the AU version is Channel Link, which, if enabled, gives you the old behavior of summing all channles together, but if disabled, gives you a new option of each channel being processed independently.
mda VST plug-ins
This archive contains the source code for the mda freeware VST plug-ins, with the least restrictive licensing I could find.
Projects are provided for Visual C++ 5, Visual C++ 8, and XCode 2. You will need a copy of the VST 2.4 SDK to compile the plug-ins, unless you are only interested in the AU versions (kindly ported by Sophia Poirier). Code for "mda SpecMeter" is included but does not compile with any recent version of VSTGUI. The GUIs for "mda Piano" and "mda ePiano" are not included for the same reason, and also because nobody really liked them.
This code is definitely not an example of how to write plug-ins! It's obvious that I didn't know much C++ when I started, and some of the optimizations might have worked on a 486 processor but are not relevant today. The code is very raw with no niceties like parameter de-zipping, but maybe you'll find some useful stuff in there. Hopefully someone will feel like tidying everything up and put the code in a more generic format that can be wrapped into any current or future plug-in format.
Paul Kellett ([email protected]), June 2008
mda VST plug-ins
Copyright (c) 2008 Paul Kellett
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