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About TAP-plugins

License and Disclaimer

TAP-plugins are released under the GNU General Public License, version 2. See the file COPYING (included in the tarball release) for details.

DISCLAIMER: The plugins described here are available to you free of charge, and you are free to examine their internal workings, modify and re-distribute them. But keep in mind that TAP-plugins is provided "AS IS", and it comes with ABSOLUTELY NO WARRANTY of any kind, either expressed or implied, included, but not limited to, the implied warranties of MERCHANTABILITY and FITNESS FOR A PARTICULAR PURPOSE. Again, see the file COPYING for details.

Announcements and Feedback

Installation

Should be pretty easy. Download and untar the archive and cd into the resulting directory tap-plugins-x.y.z. If you store your plugin .so files in a different directory than /usr/local/lib/ladspa (which is the default) then please edit the top of the Makefile first of all. There is a default place also for RDF metadata. Set it according to your own needs. Because the plugins do not require any special library (apart from the standard GNU C library), there is no ./configure script. All you have to do is issue a make and (as root) a make install. If you don't get any errors (you shouldn't), fire up your host and check out your new plugins. You may also want to check the README file for additional information.

A few words about signal routing

When connecting plugins to audio inputs/outputs and to each other in order to build a digital audio processing chain, it is important to understand how audio is actually routed between individual input/output ports. Plugins can have a different number of audio inputs and outputs, and routing is handled by the host.

The TAP-plugins have either 1 input and 1 output (mono plugins) or 2 inputs and 2 outputs (stereo plugins). No other combinations are used, although it would be perfectly legal for a plugin to have 1 input and 2 outputs, for example.

When you apply a mono plugin to a mono track, everything is clear. The one and only input of the plugin gets connected to the mono signal, and the single plugin output is routed back somewhere (to the input of another plugin, or to a master out, etc). Another simple case is when you apply a stereo (2 in / 2 out) plugin to a stereo track. Here again, everything gets connected as you would expect.

But what happens if you apply a mono plugin to a stereo track (or put it in a chain of plugins after a plugin with stereo outputs)? In this case, actually two instances are created of the same plugin, and the two input channels are processed by these two separate plugins to get two output channels. The two instances are binded to the same GUI controls, so you see and adjust only one plugin GUI. But keep in mind that it's actually two plugins running in the background, consuming twice as much memory and CPU power as one single plugin would cost. The CPU usage metrics (provided below) for mono plugins assume you are actually using only one instance, so you should double the figures when applying them to a stereo track.

About CPU usage metrics

CPU usage metrics are provided for each plugin. This is not a very accurate way to tell how much a plugin setup costs, but it should provide a general idea about how expensive each plugin is, and it is better than nothing. Please don't rely on these figures when you need to predict the system load in a situation where the results of system overload (stopping of the transport, falling out of sync, etc) are unacceptable. For example, if you are recording a live show or doing live PA work, never ever go into calculations that result in saying that you will need only 98% of your CPU power when using such-and-such plugins. This is extremely dangerous, and there are much easier ways to commit suicide as well.

The CPU usage figures were measured running the plugins within Ardour, and recording the increase of CPU usage indicated at the top of the Ardour Editor window. The machine used for measurements was a uniprocessor Pentium IV machine with a processing power of approximately 3395 BogoMIPS. Mono plugins were applied to mono tracks, and stereo plugins were applied to stereo tracks (read the above section about routing to understand why this is important). Measurements were made at both 44.1 kHz and 96 kHz sampling frequencies. CPU usage is proportional to the sampling rate (at least in theory), so at 48 kHz you should expect somewhat greater CPU demand than at 44.1 kHz, and at 192 kHz expect roughly twice as much CPU usage as the value at 96 kHz. The actual figures are provided as part of the plugin manuals.

Plugin manuals

This section contains detailed information, benchmarks and usage tips for each available plugin. It is intended to serve as a reference manual for the TAP-plugins.

TAP AutoPanner

The AutoPanner is a very well-known effect; its hardware incarnation originates in the age of voltage controlled synthesizers. Its main use is to liven up synth tracks in the mix.

General information

Usage tips

The "Depth" control sets the wideness of panning. A setting of 100% means that the mono input signal is panned between the extreme left and right positions. If you apply the plugin to a stereo track, it behaves as if you were adjusting the balance control on your home stereo (the left and right channels are not mixed together, but their gains are controlled separately). If you feel that the overall loudness of the track decreases when this effect is applied, you can compensate this by setting the "Gain" control.

Summary of user controls

Notes

This plugin was implemented as an enhancement of the TAP Tremolo plugin: processing was extended to 2 tracks, with the LFO-s phase-shifted to be in counter-phase.

TAP DeEsser

TAP DeEsser is a plugin for attenuating higher pitched frequencies in vocals such as those found in 'ess', 'shh' and 'chh' sounds. Almost any vocal recording will contain 'ess' sounds, whether a strong vocal delivery, from bad recording, speech impediments or simply many 'ess' words spoken together. Wind instruments and other musical instruments can also create shrill high-pitched noises. Audio engineers need to control these harsh 'ess' sounds in most recordings.

General information

Usage tips

Audio input is fed into the audio path and the so-called Sidechain. In the Sidechain, the signal is filtered and then an attenuation value is computed based on the filter output. This computation is done using a hard-knee compressor characteristic with a threshold level set by the "Threshold" control and a compression ratio of 1:2. The compressor attack/release time is 10 ms for instant operation.

The filter applied to the Sidechain signal has two modes of operation: Highpass and Bandpass. Highpass mode is useful for attenuating several different 'ess' sounds. Bandpass mode is more suitable for attenuating a specific high frequency. The chosen filter's frequency is set by the "Frequency" control.

General advice about setting the "Frequency" control: The sounds attenuated by this plugin can be fit into four categories: male 'ess', male 'ssh', female 'ess' and female 'ssh' sounds. 'Shh' settings are quite applicable for 'ch', 'th', and hard consonants, such as 't', 'd', and 'k' as well. The following table lists frequencies for these four types of sounds. Note that these are only provided as a starting point for your own experimentations, as the characteristic frequencies of individual vocalists can vary in a suprisingly wide range. Use your ears!

Recommended frequency values

Generally, female 'ess' and 'shh' sounds vary more in frequency than those of males. Due to this situation, you may find that using the Sidechain filter in the Highpass mode may be more responsive. Otherwise, with the Sidechain filter in Bandpass mode, only a narrow frequency area is being responded to. So if a singer has esses with varying frequency, try the Highpass mode of the Sidechain.

When choosing between Sidechain modes (e.g. Highpass or Bandpass) and setting the "Frequency" it may be very helpful to hear the Sidechain signal, which controls the compression of the through-passing audio: this is why the Monitor control is provided. While listening to the Sidechain signal, try to set the controls so that almost everything you want attenuated can be well heard, and even more importantly, almost everything you want to keep intact doesn't show up in the Sidechain sound.

Another aid in setting the plugin is the "Attenuation" output meter: this shows the momentary attenuation applied to the signal. It shouldn't show a significant attenuation all the time. Instead, it should indicate that attenuation springs into action only occassionally.

Summary of user controls

Notes

Setting the "Threshold" too low will result in an unpleasant muffled sound. Also, if the singer starts sounding as if she had suddenly lost her teeth, take this as a sure sign that your "Threshold" setting is way too low. Use the ever popular Bypass button and watch out for side effects like this.

This plugin is best on vocal tracks, although it can be applied to mixed tracks as well. However, you should expect an increased amount of side effects (especially muffling and pumping of the sound) when attempting to de-ess the vocals of a mixed track.

Internal variables that depend on user settings are re-computed only when the corresponding user settings are changed. This keeps the plugin from being hard RT capable, although the effect of this occassional re-computation is not really noticeable when looking at CPU usage measures.

Because the Sidechain processing algorithm involves converting from linear to decibel values for every sample, a custom, very fast lin2db function was implemented via a log10 table, which is computed and stored when the plugin library is loaded by the host. Without this hack (that is, calling log10f() for every sample), CPU usage would be almost exactly twice this much.

TAP Equalizer

TAP Reverberator

TAP Reverberator is unique among reverberators freely available on the Linux platform. It supports creating no less than 38 reverberation effects, but its design permits this to be extended even further by the user, without doing any actual programming.

The design is based on the comb/allpass filter model. Comb filters create early reflections and allpass filters add to this by creating a dense reverberation effect. The output of the set of comb and allpass filters (also called the reverberator chamber) is processed further by sending it through a bandpass filter. The resulting band-limited reverberation is very similar to the natural reverberation that occurs in acoustic rooms. To achieve an even more natural-sounding effect, all comb filters have high-frequency compensation in their feedback loop. This is to model that the reflexion ratio of acoustic surfaces is the function of frequency: higher frequencies are attenuated more, and thus decay time of higher frequency components is significantly shorter.

To enhance the reverberation sound even further, a special option called Enhanced Stereo is provided. When turned on (which is the default), it results in an added spatial quality. This feature is most noticeable when applying the plugin to mono tracks: the sound of these tracks will "open up" and a beautiful spatial effect will be heard.

General information

Usage tips

Despite the rather complex algorithm of this plugin, usage is relatively simple and the user is required to adjust only a limited number of global controls. This is achieved by defining "presets" which actually mean larger sets of control values. When choosing a preset, the plugin loads all values associated to that preset, and operates according to the newly loaded values.

The most important global variable of the whole effect is the decay time. By adjusting this, you can create the feel of a larger acoustic space (set the decay to more than 2-3 seconds for this), as well as a "tight", relatively dry effect (for which you should set a much lower value). But you should be aware of the fact that not every preset (reverberation type) sounds good at any decay setting (see the table below for recommended decay values). For example, a Room (Small) will not sound too good when you set the decay to a very large value, and vice versa, a Hall (Large) will not sound optimal at very small decay settings. However, when choosing a more artificial type of reverberation (Afterburn, Pulse Chamber (Reverse) or Warble Chamber, for example), there is no "natural" decay time: set it as you see fit according to the effect you are trying to achieve.

General advice about adjusting the dry and wet signal levels: the wet level should be about 3 to 15 dB lower than the dry level, according to the preset type you are using. Of course if you want to create an artificial sounding effect, you can decrease the dry level, even down to -70 dB if that is what you want. However, one thing to know is that you shouldn't set the wet level to a very high value (generally not above 0 dB) because the output level of the plugin may become too high and you may overdrive the next plugin in the chain, or you may cause signal clipping in the master output of your multitrack. Furthermore, because the wet signal is added to the incoming dry signal, you should decrease the dry level as well by a few dB-s (and then set the wet level according to this) so as not to raise the overall loudness of the track.

Usually, when trying to create a natural-sounding reverberation effect, all components (comb filters, allpass filters, the bandpass filter and the stereo enhancement mode) should be switched on. However, to create artificial effects, it is possible to turn off any of these effects. For example, if you only want the sound of multiple echoes, you can turn off everything but the comb filters; if you want to create a more unnatural, harsh effect, turn off the bandpass filter processing. (Also note that bandpass filters of different chambers have different low-pass and high-pass frequencies, since this filtering also affects the nature of the reverberated sound very much.) Naturally, components that are switched off don't consume CPU power.

As mentioned earlier, not all presets sound good with any decay setting. The following table contains decay time settings adequate for particular presets. Decay times are shown in seconds. As a rule of thumb, when decay times are above 3 seconds, the "Wet Level" should be at least 6 or more decibels below the "Dry Level". The values below are only general advice, and they are provided merely as a starting point for your experimentation. You are free to use any value that sounds good for your mix.

Recommended decay times

Q: What is that HD in the name of some presets?

A: HD stands for High Density. Presets marked with this are enhanced versions of other presets, for example Hall (Large) - HD is derived from Hall (Large). Enhancement means additional comb and/or allpass filters, which result in an even smoother reverberation effect. However, more filters mean more CPU usage, so it's up to you to decide whether to choose them or not. If CPU usage is not a great concern, it is recommended to use the HD versions where available.

Q: The GUI of this plugin looks like crap in [host-other-than-Ardour]. Why? Any workaround?

A: Unfortunately LADSPA plugins have virtually no control over what their GUI-s will look like in a particular host. This is because the host generates plugin GUI-s, based upon port definitions provided by the plugin. In the case of TAP Reverberator, it is desirable to have the names of the presets displayed to the user. This can only be achieved by putting all the preset names in a long string that is displayed by the host as a name of a plugin-defined control. It is likely that different hosts handle this situation in different ways, and the list of presets you see may be displayed in a somewhat crappy way. This is because the long string containing all the preset names has been optimized to be good-looking in Ardour. If this long string (which, in addition, contains several newline characters) causes problems with your host, or is damaging your eye, it is possible to turn it off by commenting out the line saying

#define _SHOW_PRESET_NAMES_IN_GUI_

in tap_reverb.h (it is at the top of the file, so you should find it quickly). If you do this, the text displayed next to the preset selector will say nothing but "Preset" (after a re-compile, of course). When using the plugin this way, you will have to open this manual page in a browser to see the preset names.

Summary of user controls

Notes

The maximum number of filters is 20 combs and 20 allpass filters. Actually it is double this much because every filter exists in two instances for the two channels. This amount allows for the creation of very dense, very smooth-sounding reverberation chambers. However, currently the most complex preset called Gymnasium (Bright) - HD employs 7 comb and 7 allpass filters, so there is plenty of capacity left unused in this plugin implementation.

Comb filters are implemented as first-order IIR filters with a peaking EQ in the feedback loop set to a negative gain (the exact gain value depends on the frequency response parameter of the comb filter) at 10 kHz. Allpass filters are implemented as ordinary first-order IIR filters.

When Enhanced Stereo is enabled, parameters of the two instances of the same filter are set to a slightly different value. Without this option, the two input channels are processed in the exact same way. It is absolutely recommended to turn this on when applying the plugin to mono tracks, but it is also good on stereo tracks.

Because the varying complexity of chambers and the fact that options can be switched on/off individually, CPU usage is a function of user settings. For this reason, this plugin is not hard RT capable. The CPU usage was measured with all options turned on, and the preset set to Gymnasium (Bright) - HD, which is the most complex chamber.

TAP Scaling Limiter

You want to maximize the loudness of your master tracks. Your drummer has the habit of playing with varying velocity. You want to sqeeze high transient spikes down into the bulk of the audio. You want a limiter with transparent sound, but without distortion. This is for you, then. The unique design of this innocent looking plugin results in the ability to achieve signal level limiting without audible artifacts.

Most limiters operate on the same basis as compressors: they monitor the signal level, and when it gets above a threshold level they reduce the gain on a momentary basis, resulting in an unpleasant "pumping" effect. Or even worse, they chop the signal at the top. This plugin actually scales each half-cycle individually down to a smaller level so the peak is placed exactly at the limit level. This operation (from zero-cross to zero-cross) results in an instantaneous blending of peaks and transient spikes down into the bulk of the audio.

General information

Usage tips

Although this limiter is an exceptionally clean sounding one, it is possible to introduce distortion by setting the limit level down into the bulk of the audio. Only transients, drum beats and such things (which "stand out" of the waveform) should be attacked. To illustrate the optimal setting of this plugin, let's look at the following waveforms (seemingly some unmastered rock music):

As you can see, no signal gets above the limit level, so this plugin acts as a brickwall limiter (with a compression ratio of 1:inf above the limit level). But the two waveforms above sound exactly the same! This is possible because transients last for a very short amount of time, and despite their high level, they don't carry very much energy compared to the bulk of the audio. Now let's take a closer look at the same waveforms:

As it can be seen, the waveform does not lose any of its details, because half-cycles (from zero-cross to zero-cross) are individually scaled down to the limit level (only those that have peaks above the limit, of course).

When setting the "Limit Level", you should not go more than 7-8 dB below the average peak level of the signal, which should be observed via some peak-holding level meter. When using this plugin in Ardour, this can be done very easily because the meters of the mixer strips provide an excellent way to monitor exact peak values. Assuming an unmastered mix of some rock music with drums, bass, guitars etc. you can usually get down to the point where the Ardour level meter doesn't move very much. Without hearing any distortion, of course. Because high transients are squeezed down, the overall signal level can be raised with the "Output Volume" control. Thus, this plugin is a very effective way to transparently maximize the loudness of your mixes.

Summary of user controls

Notes

This plugin has a processing function that examines the input audio signal and makes decisions based on sample values. So by definition, this plugin cannot qualify as hard RT capable, although CPU demand tends to be fairly constant during usage.

TAP Stereo Echo

This plugin supports conventional mono and stereo delays, ping-pong delays and the Haas effect (also known as Cross Delay Stereo). A relatively simple yet quite effective plugin.

General information

Usage tips

If you want to create a conventional mono or stereo delay, all you have to do is set the delay times and feedback ratios for the two input channels. If the feedback is set to zero, then only one delay of the input is created (as if a traditional tape delay was used). When feedback is greater than zero, the same bit of audio is sent back and delayed over and over again, with decreasing amplitudes. In this case the decay time is dependent of the feedback value.

When you turn the Cross Mode switch on, the feedback loops are being feeded by the delayed signal of the opposite input channel. A sound coming from the left input channel will be delayed by the Left Delay time value, but the output of the left delay ringbuffer will be sent back to the right input (instead of the left) with regard to the right channel's feedback setting. In Cross mode, a sound appearing at one input will be bouncing between the two output channels, hence the popular name "ping-pong delay". If you use this effect with high delay times and feedback values, even the dumbest listener will notice that your mix is stereo.

A third effect achievable with this plugin is the so-called Haas effect. This effect is founded by the following psycho-acoustic experience: if a sound reaches one ear, and the same sound reaches the other ear but with a time shift of 15-40 milliseconds, only one sound is heard, but with a spatial feeling. The Haas effect (also known as Cross Delay Stereo, which refers to the means by which the effect is created) is widely used by mixing engineers to avoid their mixes being "pan-potted mono", or to "stretch out" their otherwise mono guitar, vocal etc. tracks in space. This effect is a great alternative to reverberation (although it produces a noticeably different quality).

Summary of user controls

Notes

The maximum delay time (which is currently 2000 ms) can be set to a greater value in tap_echo.c if needed. When activating the plugin, memory is allocated for a ringbuffer which is large enough to contain audio as long as this value. Because this amount of memory is proportional to the maximum delay, it is not desirable to set it to a very large value if you don't want to actually use it since you will be only wasting memory. (This waste will be temporary of course, since memory is freed when the host deactivates the plugin -- that is, when you remove it from the mixer/patchbay/whatever your host has.)

TAP Tremolo

The tremolo effect is probably one of the most ancient effects, originated in the earliest days of the history of studio recording. It lost some of its popularity over time (and with the emerge of more exciting digital effects), but you still hear this effect on newer recordings from time to time.

General information

Usage tips

This plugin sounds good mainly on acoustic or electric guitar tracks, and should be used only occassionally. To create a "floating" effect, set the frequency value to 3-6 Hz. To create a more intensive effect, set it to 8-15 Hz (recommended mostly on electric guitar tracks). The depth should be probably set somewhere between 50% and 90%. If you feel that the overall loudness of the track decreases when this effect is applied, you can compensate this by setting the "Gain" control.

Summary of user controls

Notes

The sine function is implemented via a sine table, which is computed when the host loads the plugin library. This sort of implementation was necessary because calling sinf() for every sample proved to be way too expensive -- that very call itself doubled the CPU usage.