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/opt/src/beast/docs/psource/bse-objects.scad
SYNOPSIS
object
| BseAdder; |
object
| BseAmplifier; |
object
| BseArtsCompressor; |
object
| BseAtanDistort; |
object
| BseBalance; |
object
| BseBiquadFilter; |
object
| BseBus; |
object
| BseBusModule; |
object
| BseCSynth; |
object
| BseConstant; |
object
| BseContainer { |
Signals: |
void
| ::item-added
| (arg1); |
void
| ::item-remove
| (arg1, arg2); |
}; |
object
| BseContextMerger; |
object
| BseContribSampleAndHold; |
object
| BseCxxBase; |
object
| BseDataPocket { |
Signals: |
void
| ::entry-added
| (arg1); |
void
| ::entry-removed
| (arg1); |
void
| ::entry-changed
| (arg1); |
}; |
object
| BseDavBassFilter; |
object
| BseDavChorus; |
object
| BseDevice; |
object
| BseEditableSample { |
Signals: |
void
| ::changed
| (); |
}; |
object
| BseEffect; |
object
| BseFreeVerb; |
object
| BseIIRFilter; |
object
| BseInstrumentInput; |
object
| BseInstrumentOutput; |
object
| BseItem { |
Signals: |
void
| ::parasites-added
| (arg1); |
void
| ::parasite-changed
| (arg1); |
}; |
object
| BseJanitor { |
Signals: |
void
| ::progress
| (arg1); |
void
| ::action-changed
| (arg1, arg2); |
void
| ::action
| (arg1, arg2); |
void
| ::shutdown
| (); |
}; |
object
| BseLadspaModule; |
object
| BseMidiController; |
object
| BseMidiDevice; |
object
| BseMidiDeviceALSA; |
object
| BseMidiDeviceNULL; |
object
| BseMidiDeviceOSS; |
object
| BseMidiInput; |
object
| BseMidiNotifier { |
Signals: |
void
| ::midi-event
| (arg1); |
}; |
object
| BseMidiSynth; |
object
| BseMidiVoiceInput; |
object
| BseMidiVoiceSwitch; |
object
| BseMixer; |
object
| BseMult; |
object
| BseNoise; |
object
| BseObject { |
Signals: |
void
| ::release
| (); |
void
| ::icon-changed
| (); |
}; |
object
| BsePart { |
Signals: |
void
| ::range-changed
| (arg1, arg2, arg3, arg4); |
void
| ::links-changed
| (); |
}; |
object
| BsePcmDevice; |
object
| BsePcmDeviceALSA; |
object
| BsePcmDeviceNull; |
object
| BsePcmDeviceOSS; |
object
| BsePcmInput; |
object
| BsePcmOutput; |
object
| BsePcmWriter; |
object
| BseProject { |
Signals: |
void
| ::state-changed
| (arg1); |
}; |
object
| BseSNet { |
Signals: |
void
| ::port-unregistered
| (); |
}; |
object
| BseSequencer; |
object
| BseServer { |
Signals: |
void
| ::registration
| (arg1, arg2, arg3); |
void
| ::message
| (arg1); |
void
| ::script-start
| (arg1); |
void
| ::script-error
| (arg1, arg2, arg3); |
}; |
object
| BseSimpleADSR; |
object
| BseSnooper; |
object
| BseSong { |
Signals: |
void
| ::pointer-changed
| (arg1); |
}; |
object
| BseSource { |
Signals: |
void
| ::io-changed
| (); |
void
| ::probes
| (arg1); |
}; |
object
| BseStandardGusPatchEnvelope; |
object
| BseStandardOsc; |
object
| BseStandardQuantizer; |
object
| BseStandardSaturator; |
object
| BseStorage; |
object
| BseSubIPort; |
object
| BseSubOPort; |
object
| BseSubSynth; |
object
| BseSummation; |
object
| BseSuper; |
object
| BseTrack { |
Signals: |
void
| ::changed
| (); |
}; |
object
| BseWave; |
object
| BseWaveOsc { |
Signals: |
void
| ::notify-pcm-position
| (arg1, arg2); |
}; |
object
| BseWaveRepo; |
object
| DavCanyonDelay; |
object
| DavOrgan; |
object
| DavSynDrum; |
object
| DavXtalStrings; |
DESCRIPTION
BseAdder
The Adder is a very simplisitic prototype mixer that just sums up incoming signals (it does allow for switching to subtract mode though)
Channels:
audio-in1: | |
Audio Input 1 |
audio-in2: | |
Audio Input 2 |
audio-out: | |
Audio Output |
|
Properties:
subtract: | |
Use subtraction to combine samplevalues (instead of addition) |
|
BseAmplifier
BseAmplifier provides input signal controlled amplification or attenuation. It supports two control inputs and two audio inputs which are mixed together after balance and level adjustments. The gain setting controls preamplification of the mixed audio signal. The mixed control signal, weighted by a strength setting, determines additional amplification, allowing for external sources to modulate the overall volume (tremolo). The mixed control signal can influence the amplification linearly (to amplify other control signals) or exponentially (to amplify audio signals). Finally, the master volume controls amplification of the resulting output signal.
Authors: Tim Janik
License: GNU Lesser General Public License
Channels:
audio-in1: | |
First audio input |
audio-in2: | |
Second audio input |
ctrl-in1: | |
First control input |
ctrl-in2: | |
Second control input |
audio-out: | |
Amplified audio output |
|
Properties:
alevel1: | |
Attenuate the level of audio input 1 (Range: 0 .. 100) |
alevel2: | |
Attenuate the level of audio input 2 (Range: 0 .. 100) |
abalance: | |
Balance audio input levels (Range: -100 .. 100) |
clevel1: | |
Attenuate the level of control input 1 (Range: 0 .. 100) |
clevel2: | |
Attenuate the level of control input 2 (Range: 0 .. 100) |
cbalance: | |
Balance control input levels (Range: -100 .. 100) |
ctrl_mul: | |
Switch the two control inputs to work like a chain of gain controls, rather than mixing them together, weighted by balance. |
ctrl_exp: | |
Toggle exponential and linear control response |
ostrength: | |
Adjust how much the control signals affect the output volume (Range: 0 .. 100) |
base_level: | |
Base amplification (the control signal adds up to this) (Range: 0 .. 100) |
master_volume: | |
Output stage amplification in decibel (Range: 0.0000158 .. 15.8489319) |
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BseArtsCompressor
ArtsCompressor compresses the sound.
Compression is a dynamic operation that consists of two parts:
( 1) the current input volume of the signal is detected
( 2) if it exceeds a given theshold, the volume of the output signal will be reduced
The input volume detection has an attack and a release half-life time which can be specified - in milliseconds - with the corresponding properties. This envelope causes the compressor to adapt slowly to the actual input signal level.
The ratio specifies how the volume of a signal should be reduced, if it exceeds the threshold. A compression ratio of 2: 1 means for instance that if the input volume is 2 dB over the threshold, the output volume will be 1 dB over the threshold. Example (threshold = -10, ratio 2: 1):
input = -20 dB => output = -20 dB
input = -10 dB => output = -10 dB
input = 0 dB => output = -5 dB
input = 10 dB => output = 0 dB
input = 20 dB => output = 5 dB
Compression is often thought off as an operation to make music "sound louder". To achieve this, the first step is to reduce the dynamic range like in the example above. As the loud parts of the music have been reduced in volume, we can now amplify everything, without risking distortion or clipping. This has the overall effect of the music sounding louder. In our example, an output amplification of 5 dB would be okay, if the input signal never exceeded 0 dB.
Authors: Matthias Kretz, Stefan Westerfeld
License: GNU General Public License
Channels:
audio-in1: | |
Audio input 1 |
audio-in2: | |
Audio input 2 |
audio-out1: | |
Compressed audio output 1 |
audio-out2: | |
Compressed audio output 2 |
|
Properties:
attack: | |
Set the attack time in milliseconds (Range: 0.1 .. 250) |
release: | |
Set the release time in milliseconds (Range: 0.1 .. 250) |
threshold_db: | |
Only parts louder than threshold are compressed (Range: -100 .. 0) |
ratio_to_one: | |
Set the compression ratio to x:1 (Range: 1 .. 20) |
output_db: | |
Set the output amplification (Range: -20 .. 20) |
auto_output: | |
Adjust the output amplification automatically according to threshold and ratio |
|
BseAtanDistort
BseAtanDistort compresses or expands the input signal with distortion (in a manner similar to the atan( 3) mathematical function, thus it's name). The strength with which the input signal is treated is adjustable from maximum attenuation to maximum boost.
Channels:
audio-in: | |
Audio Input Signal |
audio-out: | |
Distorted Audio Output |
|
Properties:
boost_amount: | |
The atan distortion boost amount (strength) ranges from maximum attenuation (0%) to maximum boost (100%). (Range: 0 .. 100) |
|
BseBalance
BseBalance is a stereo panning module with two audio and two control inputs, each of which can be seperately level adjusted. The resulting output balance is controlled by a balance adjustment and the sum of the control inputs. To avoid aliasing artefacts for rapid balance changes, the control signal is preprocessed by a simple lowpass filter with adjustable cutoff frequency.
Authors: Tim Janik
License: GNU Lesser General Public License
Channels:
audio-in1: | |
First audio input |
audio-in2: | |
Second audio input |
ctrl-in1: | |
First control input |
ctrl-in2: | |
Second control input |
left-out: | |
Left audio output |
mix-out: | |
Mixed audio output |
right-out: | |
Right audio output |
|
Properties:
alevel1: | |
Attenuate the level of audio input 1 (Range: 0 .. 100) |
alevel2: | |
Attenuate the level of audio input 2 (Range: 0 .. 100) |
abalance: | |
Balance audio input levels (Range: -100 .. 100) |
clevel1: | |
Attenuate the level of control input 1 (Range: 0 .. 100) |
clevel2: | |
Attenuate the level of control input 2 (Range: 0 .. 100) |
cbalance: | |
Balance control input levels (Range: -100 .. 100) |
ostrength: | |
Adjust how much the control signals affect the output balance (Range: 0 .. 100) |
lowpass: | |
Lowpass filter frequency for the control signal (Range: 100 .. 1000) |
obalance: | |
Adjust output balance between left and right (Range: -100 .. 100) |
|
BseBiquadFilter
BseBiquadFilter - an infinite impulse response filter, consisting of 2-zeros/2-poles segments (thus it's name: bi-quadratic). Various types of filters are supported with modulatable center (Cut-Off) frequency and gain input signals.
Channels:
audio-in: | |
Unfiltered Audio Signal |
freq-in: | |
Center Frequency Input |
freq-mod-in: | |
Frequency Modulation Input |
gain-mod-in: | |
Gain Modulation Input |
audio-out: | |
Filtered Audio Signal |
|
Properties:
filter_type: | |
The filter design type |
freq: | |
Filter cutoff frequency in Hertz (Range: 0.00005 .. 19999.9) |
note: | |
Filter cutoff frequency as note, converted to Hertz according to the current musical tuning (Range: 0 .. 131) |
gain: | |
Gain [dB] (Range: -48 .. 48) |
fm_perc: | |
Strength of linear frequency modulation (Range: 0 .. 100) |
exponential_fm: | |
Perform exponential frequency modulation instead of linear |
fm_n_octaves: | |
Number of octaves to be affected by exponential frequency modulation (Range: 0 .. 5) |
gain_perc: | |
Strength of gain modulation (Range: 0 .. 100) |
|
BseBus
Bus implementation for songs, used to route track audio signals to the master output.
Channels:
left-audio-in: | |
Left channel input |
right-audio-in: | |
Right channel input |
left-audio-out: | |
Left channel output |
right-audio-out: | |
Right channel output |
|
Properties:
inputs: | |
Synthesis signals (from tracks and busses) used as bus input |
outputs: | |
Mixer busses used as output for synthesis signals |
mute: | |
Mute: turn off the bus volume |
solo: | |
Solo: mute all other busses |
sync: | |
Syncronize left and right volume |
left_volume: | |
Volume adjustment in decibel of left bus channel (Range: 0.0000158 .. 15.8489319) |
right_volume: | |
Volume adjustment in decibel of right bus channel (Range: 0.0000158 .. 15.8489319) |
|
BseBusModule
Synthesis module used internally by BseBus
Authors: Tim Janik
License: GNU Lesser General Public License
Channels:
audio-in1: | |
First audio input |
audio-in2: | |
Second audio input |
audio-out1: | |
First audio output |
audio-out2: | |
Second audio output |
|
Properties:
volume1: | |
(Range: 0 .. 1000) |
volume2: | |
(Range: 0 .. 1000) |
|
BseCSynth
BSE Synthesis (Filter) Network
BseConstant
This module provides constant signal outputs
Channels:
const-out1: | |
Constant Output 1 |
const-out2: | |
Constant Output 2 |
const-out3: | |
Constant Output 3 |
const-out4: | |
Constant Output 4 |
|
Properties:
value_1: | |
Constant signal value (Range: -1 .. 1) |
frequency_1: | |
Constant signal value interpreted as frequency value in Hertz (Range: 0 .. 24000) |
note_1: | |
Constant signal value as note, converted to Hertz according to the current musical tuning (Range: 0 .. 131) |
value_2: | |
Constant signal value (Range: -1 .. 1) |
frequency_2: | |
Constant signal value interpreted as frequency value in Hertz (Range: 0 .. 24000) |
note_2: | |
Constant signal value as note, converted to Hertz according to the current musical tuning (Range: 0 .. 131) |
value_3: | |
Constant signal value (Range: -1 .. 1) |
frequency_3: | |
Constant signal value interpreted as frequency value in Hertz (Range: 0 .. 24000) |
note_3: | |
Constant signal value as note, converted to Hertz according to the current musical tuning (Range: 0 .. 131) |
value_4: | |
Constant signal value (Range: -1 .. 1) |
frequency_4: | |
Constant signal value interpreted as frequency value in Hertz (Range: 0 .. 24000) |
note_4: | |
Constant signal value as note, converted to Hertz according to the current musical tuning (Range: 0 .. 131) |
|
BseContainer
Base type to manage BSE items
BseContextMerger
Internal CONTEXT Voice glue object (merger)
BseContribSampleAndHold
SampleAndHold samples the input signal when triggered and holds it until triggered again.
Authors: Artem Popov
License: GNU General Public License
Channels:
audio-in: | |
Audio Input |
trigger-in: | |
Trigger Input |
audio-out: | |
Audio Output |
|
BseDavBassFilter
DavBassFilter is a low-pass resonant TB-303 style filter based on the VCF303 portions of gsyn v0.2. The filter algorithm is partly based on the one described in "Musical Applications of Microprocessors" by Hal Chamberlin.
Authors: David A. Bartold, Tim Janik
License: GNU Lesser General Public License
Channels:
audio-in: | |
Audio input |
trigger-in: | |
Trigger input |
audio-out: | |
Filtered audio output |
|
Properties:
cutoff_perc: | |
Set the cutoff frequency percentage (Range: 0 .. 100) |
reso_perc: | |
Set the amount of resonance in percent (Range: 0 .. 100) |
env_mod: | |
Set the envelope magnitude in percent (Range: 0 .. 100) |
env_decay: | |
Set the decay length in percent (Range: 0 .. 100) |
trigger: | |
Trigger filter |
|
BseDavChorus
DavChorus adds more depth to sounds
Authors: David A. Bartold
License: GNU Lesser General Public License
Channels:
audio-in: | |
Audio input |
audio-out: | |
Chorus audio output |
|
Properties:
wet_out: | |
Set the amount of modified data to mix (Range: 0 .. 100) |
|
BseDevice
Abstract device base type
BseEffect
BseEffect implements an abstract C++ effect base.
BseFreeVerb
BseFreeVerb - Free, studio-quality reverb (SOURCE CODE in the public domain) Written by Jezar at Dreampoint - http://www.dreampoint.co.uk
Channels:
left-audio-in: | |
Left Input |
right-audio-in: | |
Right Input |
left-audio-out: | |
Left Output |
right-audio-out: | |
Right Output |
|
Properties:
room_size: | |
Room Size (Range: 0.7 .. 0.98) |
damping: | |
Damping [%] (Range: 0 .. 100) |
wet_level: | |
Wet Level [dB] (Range: 0 .. 3) |
dry_level: | |
Dry Level [dB] (Range: 0 .. 2) |
width: | |
Width [%] (Range: 0 .. 100) |
|
BseIIRFilter
BseIIRFilter is an infinite impulse response filter of variable order
Channels:
audio-in: | |
Unfiltered Input |
audio-out: | |
Filtered Output |
|
Properties:
filter_algorithm: | |
The filter design type |
filter_type: | |
The type of filter to use |
order: | |
Order of Filter (Range: 1 .. 18) |
epsilon: | |
Passband falloff at cutoff frequency (Range: 0 .. 0.98) |
cut_off_freq: | |
Filter cutoff frequency (Range: 0.00005 .. 20000) |
cut_off_note: | |
Filter cutoff frequency as note, converted to Hertz according to the current musical tuning (Range: 0 .. 131) |
cut_off_freq_2: | |
Second filter cutoff frequency (Range: 0.00005 .. 20000) |
cut_off_note_2: | |
Filter cutoff frequency as note, converted to Hertz according to the current musical tuning (Range: 0 .. 131) |
|
BseInstrumentInput
Virtual input module for synthesis networks which implement instruments
Channels:
frequency: | |
Note Frequency |
gate: | |
High if the note is currently being pressed |
velocity: | |
Velocity of the note press |
aftertouch: | |
Velocity while the note is pressed |
|
BseInstrumentOutput
Virtual output module for synthesis networks which implement instruments
Channels:
left-audio: | |
Left Channel Output |
right-audio: | |
Right Channel Output |
synth-done: | |
High indicates the instrument is done synthesizing |
|
BseItem
Base type for objects managed by a container
BseJanitor
BSE connection interface object
Properties:
ident: | |
Script Identifier |
connected: | |
Connected |
status_message: | |
Status Message |
exit_code: | |
Exit Code (Range: -256 .. 256) |
exit_reason: | |
Exit Reason |
|
BseMidiController
MIDI controller input module. With this module, MIDI control signals can be used in synthesis networks.
Channels:
ctrl-out1: | |
MIDI Signal 1 |
ctrl-out2: | |
MIDI Signal 2 |
ctrl-out3: | |
MIDI Signal 3 |
ctrl-out4: | |
MIDI Signal 4 |
|
Properties:
midi_channel: | |
Input MIDI channel, 0 uses network's default channel (Range: 0 .. 99) |
control_1: | |
Signal 1 |
control_2: | |
Signal 2 |
control_3: | |
Signal 3 |
control_4: | |
Signal 4 |
|
BseMidiDeviceALSA
MIDI driver implementation for the Advanced Linux Sound Architecture (http://alsa-project.org/alsa-doc/alsa-lib/midi.html)
BseMidiDeviceNULL
MIDI device implementation that does nothing
BseMidiDeviceOSS
MIDI device implementation for OSS Lite /dev/midi*
BseMidiInput
Monophonic MIDI input module. With this module, monophonic keyboard control signals can be used in synthesis networks.
Channels:
frequency: | |
Note Frequency |
gate: | |
High if the note is currently being pressed |
velocity: | |
Velocity of the note press |
aftertouch: | |
Velocity while the note is pressed |
|
Properties:
midi_channel: | |
Input MIDI channel, 0 uses network's default channel (Range: 0 .. 99) |
|
BseMidiSynth
BSE Midi Synthesizer
Properties:
midi_channel: | |
MIDI Channel (Range: 1 .. 99) |
n_voices: | |
Maximum number of voices for simultaneous playback (Range: 1 .. 256) |
snet: | |
Synthesis network to be used as MIDI instrument |
pnet: | |
Synthesis network to be used as postprocessor |
volume_d_b: | |
Master [dB] (Range: -144 .. 24) |
volume_perc: | |
Master [%] (Range: 0 .. 1584) |
|
BseMidiVoiceInput
Internal MIDI Voice glue object (input)
BseMidiVoiceSwitch
Internal MIDI Voice glue object (switch)
BseMixer
The Mixer module sums up incomming signals, and allowes for fine adjusted weighting (volume setting) of the input sources
Channels:
audio-in1: | |
Input Channel 1 |
audio-in2: | |
Input Channel 2 |
audio-in3: | |
Input Channel 3 |
audio-in4: | |
Input Channel 4 |
audio-out: | |
Sole Output |
|
Properties:
master_volume_d_b: | |
Master [dB] (Range: -144 .. 24) |
master_volume_perc: | |
Master [%] (Range: 0 .. 1584) |
volume_d_b1: | |
Channel1 [dB] (Range: -144 .. 24) |
volume_perc1: | |
Channel1 [%] (Range: 0 .. 1584) |
volume_d_b2: | |
Channel2 [dB] (Range: -144 .. 24) |
volume_perc2: | |
Channel2 [%] (Range: 0 .. 1584) |
volume_d_b3: | |
Channel3 [dB] (Range: -144 .. 24) |
volume_perc3: | |
Channel3 [%] (Range: 0 .. 1584) |
volume_d_b4: | |
Channel4 [dB] (Range: -144 .. 24) |
volume_perc4: | |
Channel4 [%] (Range: 0 .. 1584) |
|
BseMult
Mult is a channel multiplier for ring-modulating incoming signals
Channels:
audio-in1: | |
Audio Input 1 |
audio-in2: | |
Audio Input 2 |
audio-in3: | |
Audio Input 3 |
audio-in4: | |
Audio Input 4 |
audio-out: | |
Audio Output |
|
BseNoise
Noise is a generator of (supposedly) white noise
Authors: Tim Janik
License: GNU Lesser General Public License
Channels:
BseObject
BSE Object Hierarchy base type
Properties:
uname: | |
Unique name of this object |
blurb: | |
Free form comment or description |
|
BsePart
BSE part type
Properties:
n_channels: | |
Channels (Range: 1 .. 4132) |
last_tick: | |
Last Tick (Range: 0 .. 2147483647) |
|
BsePcmDeviceALSA
PCM driver implementation for the Advanced Linux Sound Architecture (http://alsa-project.org/alsa-doc/alsa-lib/pcm.html)
BsePcmDeviceNull
Null PCM device implementation
BsePcmDeviceOSS
PCM device implementation for OSS Lite /dev/dsp
BsePcmInput
Stereo PCM sound input module, per default, signals from this module originate from recording on the standard soundcard
Channels:
left-audio-out: | |
Left channel output |
right-audio-out: | |
Right channel output |
|
Properties:
gain_volume_d_b: | |
Input Gain [dB] (Range: -144 .. 24) |
gain_volume_perc: | |
input Gain [%] (Range: 0 .. 1584) |
|
BsePcmOutput
Stereo PCM sound output module, per default, signals routed into this module are played back on the standard soundcard
Channels:
left-audio-in: | |
Left channel input |
right-audio-in: | |
Right channel Input |
|
Properties:
master_volume_d_b: | |
Master [dB] (Range: -144 .. 24) |
master_volume_perc: | |
Master [%] (Range: 0 .. 1584) |
|
BseSNet
BSE Synthesis (Filter) Network
Properties:
auto_activate: | |
Automatic activation only needs to be enabled for synthesis networks that don't use virtual ports for their input and output |
|
BseSequencer
The Sequencer produces a frequency signal according to a sequence of notes
Channels:
freq-out: | |
Frequency Signal |
note-sync: | |
Note Sync Signal |
|
Properties:
notes: | |
Notes |
length: | |
Length (Range: 1 .. 128) |
transpose: | |
Transpose (Range: -36 .. 36) |
counter: | |
Timing [ms] (Range: 0 .. 1000) |
|
BseServer
BSE Server type
Properties:
wave_file: | |
Name of the WAVE file used for recording BSE sound output |
log_messages: | |
Log messages through the log system |
|
BseSimpleADSR
Simple ADSR envelope generator
Channels:
gate-in: | |
Gate input (activates/deactivates envelope) |
retrigger-in: | |
Retrigger input (raising edge retriggers envelope) |
ctrl-out: | |
Attack-Decay-Sustain-Release envelope output |
done-out: | |
This signal goes high after the release phase has completed |
|
Properties:
attack_time: | |
Attack Time [%] (Range: 0 .. 100) |
decay_time: | |
Decay Time [%] (Range: 0 .. 100) |
sustain_level: | |
Sustain Level [%] (Range: 0 .. 100) |
release_time: | |
Release Time [%] (Range: 0 .. 100) |
time_range: | |
Time ranges in seconds: 0.5 10.0 200.0 |
|
BseSnooper
The Snooper module prints statistics about the incoming signal
Channels:
Properties:
context_id: | |
If the snooper module is created multiple times, this is the context id, which is used to actually snoop data. (Range: 0 .. 65535) |
|
BseSong
BSE Song type
Properties:
musical_tuning: | |
The tuning system which specifies the tones or pitches to be used. Due to the psychoacoustic properties of tones, various pitch combinations can sound "natural" or "pleasing" when used in combination, the musical tuning system defines the number and spacing of frequency values applied. |
tpqn: | |
Number of ticks per quarter note (Range: 384 .. 384) |
numerator: | |
Measure numerator (Range: 1 .. 256) |
denominator: | |
Measure denominator, must be a power of 2 (Range: 1 .. 256) |
bpm: | |
Beats per minute (Range: 1 .. 1024) |
pnet: | |
Synthesis network to be used as postprocessor |
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BseSource
Base type for sound sources
BseStandardGusPatchEnvelope
StandardGusPatchEnvelope is an Envelope for GUS Patches
GUS Patches contain different volume settings and tremolo settings for different notes, and the envelopes are of a non-standard form, thus using a BseWaveOsc alone is not enough to produce the sound of the GUS Patches; to accomplish the playing, you thus need to connect one (or for stereo two) BseWaveOsc(s) and their Gate Output Signal to this module.
Authors: Stefan Westerfeld
License: GNU Lesser General Public License
Channels:
frequency: | |
Frequency Input |
gate-in: | |
Gate input (activates/deactivates envelope) |
retrigger-in: | |
Retrigger input (raising edge retriggers envelope) |
audio-in: | |
Audio Input of the WaveOsc playing the patch |
audio-gate: | |
Gate Output of the WaveOsc playing the patch |
audio-out1: | |
Envelope audio output 1 (left output signal) |
audio-out2: | |
Envelope audio output 2 (right output signal) |
done-out: | |
This signal goes high after the release phase has completed |
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BseStandardOsc
StandardOsc is the BSE basis oscillator which supports various types of wave forms and modulation inputs.
Channels:
freq-in: | |
Oscillating Frequency Input |
freq-mod-in: | |
Frequency Modulation Input |
pwm-in: | |
Pulse Width Modulation Input |
sync-in: | |
Syncronization Input |
audio-out: | |
Oscillated Output |
sync-out: | |
Syncronization Output |
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Properties:
wave_form: | |
Oscillator wave form |
phase: | |
Initial phase of the oscillator wave form (cycle offset in degree) (Range: -180 .. 180) |
base_freq: | |
Oscillator frequency in Hertz, i.e. the number of oscillator cycles per second (Range: 0.00005 .. 20000) |
base_note: | |
Oscillator frequency as note, converted to Hertz according to the current musical tuning (Range: 0 .. 131) |
transpose: | |
Transposition of the oscillator frequency in semitones (Range: -72 .. 72) |
fine_tune: | |
Amount of detuning in cent (hundredth part of a semitone) (Range: -100 .. 100) |
fm_perc: | |
Strength of linear frequency modulation (Range: 0 .. 100) |
exponential_fm: | |
Perform exponential frequency modulation instead of linear |
fm_n_octaves: | |
Number of octaves to be affected by exponential frequency modulation (Range: 0 .. 5) |
self_perc: | |
Strength of self modulation (Range: 0 .. 100) |
pulse_width: | |
Proportion of the positive component duration of the pulse wave form (Pulse has to be selected as wave form for this to take effect) (Range: 0 .. 100) |
pulse_mod_perc: | |
Strength of pulse width modulation input (Pulse has to be selected as wave form for this to take effect) (Range: 0 .. 100) |
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BseStandardQuantizer
The Quantizer plugin quantizes the input signals according to a configurable step setting. This simulates low quality digital signal transfer paths or ancient computer hardware.
Authors: Tim Janik
Channels:
audio-in1: | |
Audio Input 1 |
audio-in2: | |
Audio Input 2 |
audio-out1: | |
Audio Output 1 |
audio-out2: | |
Audio Output 2 |
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Properties:
qsteps: | |
The number of different steps the output signal is quantized to. (Range: 1 .. 4294967296) |
qstep_powers: | |
The number of bits the quantization steps correspond to. (Range: 0 .. 32) |
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BseStandardSaturator
This is 2-channel saturation module. Various saturation types are supported. The 'Level' parameter adjusts the maximum saturation stage output for a normalized input signal. Some saturartion types simply clip signals beyond this threshold, while others (notably TANH and ATAN) allow overdrive signals to pass through the saturation stage up to a certain amount. The output volume allows for extra amplification after the saturation stage. For saturation types that allow overdrive signals past the saturation stage, an output volume greater than 0dB may result in overdrive signals as output of this module.
Authors: Tim Janik
Channels:
audio-in1: | |
First Audio Input |
audio-in2: | |
Second Audio Input |
audio-out1: | |
First Audio Output |
audio-out2: | |
Second Audio Output |
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Properties:
saturation: | |
The type of saturation to be used |
level: | |
Normalization level of the saturation curve in percent, some saturation types still allow overdrive signals beyond this level (Range: 0 .. 100) |
auto_output: | |
Automatically adapt the output volume according to changes in saturation level |
output_volume: | |
Output stage amplification in decibel (Range: 1 .. 125.8925412) |
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BseStorage
Storage object for item serialization
BseSubIPort
Virtual input port connector, used to provide a synthesis network with input signals from other synthesis networks
Properties:
in_port_1: | |
The port name is a unique name to establish input<->output port relationships |
in_port_2: | |
The port name is a unique name to establish input<->output port relationships |
in_port_3: | |
The port name is a unique name to establish input<->output port relationships |
in_port_4: | |
The port name is a unique name to establish input<->output port relationships |
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BseSubOPort
Virtual output port connector, used to provide a synthesis network with output signals from other synthesis networks
Properties:
out_port_1: | |
The port name is a unique name to establish input<->output port relationships |
out_port_2: | |
The port name is a unique name to establish input<->output port relationships |
out_port_3: | |
The port name is a unique name to establish input<->output port relationships |
out_port_4: | |
The port name is a unique name to establish input<->output port relationships |
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BseSubSynth
This module encapsulates whole synthesizer networks, by interfacing to/from their virtual input and output ports
Properties:
snet: | |
Synthesis network to use as embedded sub network |
in_port_1: | |
Output port name to interface from |
out_port_1: | |
Input port name to interface to |
in_port_2: | |
Output port name to interface from |
out_port_2: | |
Input port name to interface to |
in_port_3: | |
Output port name to interface from |
out_port_3: | |
Input port name to interface to |
in_port_4: | |
Output port name to interface from |
out_port_4: | |
Input port name to interface to |
in_port_5: | |
Output port name to interface from |
out_port_5: | |
Input port name to interface to |
in_port_6: | |
Output port name to interface from |
out_port_6: | |
Input port name to interface to |
in_port_7: | |
Output port name to interface from |
out_port_7: | |
Input port name to interface to |
in_port_8: | |
Output port name to interface from |
out_port_8: | |
Input port name to interface to |
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BseSummation
The Summation module adds up all inputs connected to its multi-connect input streams. The sum of all inputs connected to 'Audio In1' is available at 'Audio Sum1', and similarly, 'Audio Sum2' is the accumulatiuon signal of all streams connected to 'Audio In2'. A difference signal between 'Audio Sum1' and 'Audio Sum2' is also generated and is available at 'Audio Diff'.
Authors: Tim Janik
License: GNU Lesser General Public License
Channels:
audio-in1: | |
Audio input 1 |
audio-in2: | |
Audio input 2 |
audio-out1: | |
Accumulated audio output 1 |
audio-diff: | |
Difference between audio outputs 1 and 2 |
audio-out2: | |
Accumulated audio output 2 |
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BseSuper
Base type for item managers
Properties:
author: | |
Person changing or creating this object |
license: | |
Copyright license applying to this object |
creation_time: | |
Creation Time |
modification_time: | |
Last modification time |
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BseTrack
BSE track type
Properties:
muted: | |
Muted |
snet: | |
Synthesis network to be used as instrument |
wave: | |
Wave to be used as instrument |
midi_channel: | |
Midi channel assigned to this track, 0 uses internal per-track channel (Range: 0 .. 99) |
n_voices: | |
Maximum number of voices for simultaneous playback (Range: 1 .. 256) |
pnet: | |
Synthesis network to be used as postprocessor |
outputs: | |
Mixer busses used as output for this track |
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BseWave
BSE wave type
Properties:
locator_set: | |
Locator Set |
file_name: | |
File Name |
wave_name: | |
Wave Name |
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BseWaveOsc
BseWaveOsc is a wave based oscillator module. It plays waves at arbitrary frequency with little to no aliasing artefacts by using a tschbyscheff type II resampling filter. The plaback frequency can be specified through a frequency input, and be modulated by another control signal with linear or exponential frequency response.
Channels:
freq-in: | |
Frequency Input |
sync-in: | |
Syncronization Input |
mod-in: | |
Modulation Input |
audio-out: | |
Wave Output |
gate-out: | |
Gate Output |
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Properties:
wave: | |
Wave used as oscillator source |
channel: | |
The audio channel to play, usually 1 is left, 2 is right (Range: 1 .. 256) |
fm_perc: | |
Modulation Strength for linear frequency modulation (Range: 0 .. 100) |
exponential_fm: | |
Perform exponential frequency modulation instead of linear |
fm_n_octaves: | |
Number of octaves to be affected by exponential frequency modulation (Range: 0 .. 3) |
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DavCanyonDelay
DavCanyonDelay adds deep and long canyon-alike echos to stereo signals.
Channels:
left-in: | |
Left Audio Input |
right-in: | |
Right Audio Input |
left-out: | |
Left Audio Output |
right-out: | |
Right Audio Output |
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Properties:
left_to_right_time: | |
The time for the left to right delay (Range: 0.01 .. 0.99) |
left_to_right_feedback: | |
The feedback amount; a negative feedback inverts the signal (Range: -100 .. 100) |
right_to_left_time: | |
The time for the right to left delay (Range: 0.01 .. 0.99) |
right_to_left_feedback: | |
Set the feedback amount; a negative feedback inverts the signal (Range: -100 .. 100) |
filter_freq: | |
Reflection cutoff frequency (Range: 0.00005 .. 20000) |
filter_note: | |
Filter cutoff frequency as note, converted to Hertz according to the current musical tuning (Range: 0 .. 131) |
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DavOrgan
DavOrgan is a modifiable additive organ synthesizer
Channels:
freq-in: | |
Frequency Input |
audio-out: | |
Organ Output |
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Properties:
base_freq: | |
Organ frequency in Hertz, i.e. the pitch of the base tone (Range: 0.00005 .. 20000) |
base_note: | |
Organ frequency as note, converted to Hertz according to the current musical tuning (Range: 0 .. 131) |
transpose: | |
Transposition of the frequency in semitones (Range: -72 .. 72) |
fine_tune: | |
Amount of detuning in cent (hundredth part of a semitone) (Range: -100 .. 100) |
harm0: | |
16th Harmonic (Range: 0 .. 100) |
harm1: | |
8th Harmonic (Range: 0 .. 100) |
harm2: | |
5 1/3rd Harmonic (Range: 0 .. 100) |
harm3: | |
4th Harmonic (Range: 0 .. 100) |
harm4: | |
2 2/3rd Harmonic (Range: 0 .. 100) |
harm5: | |
2nd Harmonic (Range: 0 .. 100) |
brass: | |
Changes the organ to sound more brassy |
flute: | |
Adds flute sounds |
reed: | |
Adds reeds sound |
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DavSynDrum
DavSynDrum produces synthesized drums. It accepts the drum frequency as input channel or parameter setting. Drums are triggered through a trigger parameter or via a trigger input channel which detects raising edges. The initial frequency shift is controllable through the "Ratio In" input channel, and adjustable through a parameter.
Channels:
freq-in: | |
Drum Frequency Input |
ratio-in: | |
Frequency shift ratio (assumed 1.0 if not connected) |
trigger-in: | |
Hit the drum on raising edges |
audio-out: | |
SynDrum Output |
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Properties:
base_freq: | |
Drum frequency in Hertz (Range: 0.00005 .. 20000) |
base_note: | |
Drum frequency as note, converted to Hertz according to the current musical tuning (Range: 0 .. 131) |
trigger_vel: | |
The velocity of the drum hit (Range: 0 .. 1000) |
force_trigger: | |
Manual trigger for the drum |
res: | |
The resonance half life in number of milli seconds (Range: 1 .. 1000) |
ratio: | |
The ratio of frequency shift. (i.e. 1.0 means shift equal to the drum's base frequency) (Range: 0 .. 10) |
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DavXtalStrings
DavXtalStrings is a plucked string synthesizer, using the Karplus-Strong Algorithm. Commercial use of this module until 2004 requires a license from Stanford University.
Channels:
freq-in: | |
Pluck frequency input |
trigger-in: | |
Pluck strings on raising edge |
audio-out: | |
XtalStrings Output |
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Properties:
base_freq: | |
String oscillation frequency in Hertz (Range: 0.00005 .. 20000) |
base_note: | |
String oscillation frequency as note, converted to Hertz according to the current musical tuning (Range: 0 .. 131) |
transpose: | |
Transposition of the frequency in semitones (Range: -72 .. 72) |
fine_tune: | |
Amount of detuning in cent (hundredth part of a semitone) (Range: -100 .. 100) |
trigger_vel: | |
Velocity of the string pluck (Range: 0 .. 100) |
trigger_pulse: | |
Pluck the string |
note_decay: | |
Note decay is the 'half-life' of the note's decay in seconds (Range: 0.001 .. 4) |
tension_decay: | |
Tension of the string (Range: 0.001 .. 1) |
metallic_factor: | |
Metallicness of the string (Range: 0 .. 100) |
snap_factor: | |
Snappiness of the string (Range: 0 .. 100) |
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