diff --git a/_karplus_string_8h.html b/_karplus_string_8h.html
index 6c42153d..77a99a66 100644
--- a/_karplus_string_8h.html
+++ b/_karplus_string_8h.html
@@ -86,15 +86,14 @@
#include "Dynamics/crossfade.h"
#include "Utility/dcblock.h"
#include "Utility/delayline.h"
-#include "Filters/svf.h"
-#include "Filters/tone.h"
+#include "Filters/onepole.h"
Go to the source code of this file.
Classes | |
class | daisysp::String |
Comb filter / KS string. More... | |
Comb filter / KS string. More... | |
diff --git a/_karplus_string_8h_source.html b/_karplus_string_8h_source.html
index 30f9a6d2..7679d522 100644
--- a/_karplus_string_8h_source.html
+++ b/_karplus_string_8h_source.html
@@ -83,85 +83,93 @@
KarplusString.h |
- DaisySP
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#include <stdint.h>
#include <math.h>
Go to the source code of this file.
--Classes | |
class | daisysp::Allpass |
-Namespaces | |
namespace | daisysp |
FIR Filter implementation, generic and ARM CMSIS DSP based. | |
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This is the complete list of members for daisysp::ATone, including all inherited members.
-ATone() (defined in daisysp::ATone) | daisysp::ATone | inline |
GetFreq() | daisysp::ATone | inline |
Init(float sample_rate) | daisysp::ATone | |
Process(float &in) | daisysp::ATone | |
SetFreq(float &freq) | daisysp::ATone | inline |
~ATone() (defined in daisysp::ATone) | daisysp::ATone | inline |
- DaisySP
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#include <atone.h>
-Public Member Functions | |
void | Init (float sample_rate) |
float | Process (float &in) |
void | SetFreq (float &freq) |
float | GetFreq () |
A first-order recursive high-pass filter with variable frequency response. Original Author(s): Barry Vercoe, John FFitch, Gabriel Maldonado
-Year: 1991
-Original Location: Csound – OOps/ugens5.c
-Ported from soundpipe by Ben Sergentanis, May 2020
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- -inline | -
get current frequency
Initializes the ATone module.
sample_rate | - The sample rate of the audio engine being run. |
Processes one sample through the filter and returns one sample.
in | - input signal |
Sets the cutoff frequency or half-way point of the filter.
freq | - frequency value in Hz. Range: Any positive value. |
- DaisySP
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This is the complete list of members for daisysp::Allpass, including all inherited members.
-Allpass() (defined in daisysp::Allpass) | daisysp::Allpass | inline |
Init(float sample_rate, float *buff, size_t size) | daisysp::Allpass | |
Process(float in) | daisysp::Allpass | |
SetFreq(float looptime) | daisysp::Allpass | |
SetRevTime(float revtime) | daisysp::Allpass | inline |
~Allpass() (defined in daisysp::Allpass) | daisysp::Allpass | inline |
- DaisySP
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#include <allpass.h>
-Public Member Functions | |
void | Init (float sample_rate, float *buff, size_t size) |
float | Process (float in) |
void | SetFreq (float looptime) |
void | SetRevTime (float revtime) |
Allpass filter module
- Passes all frequencies at their original levels, with a phase shift.
- Ported from soundpipe by Ben Sergentanis, May 2020
void Allpass::Init | -( | -float | sample_rate, | -
- | - | float * | buff, | -
- | - | size_t | size ) | -
Initializes the allpass module. -\param sample_rate The sample rate of the audio engine being run. -
buff | Buffer for allpass to use. |
size | Size of buff. |
in | Input sample. |
Sets the filter frequency (Implemented by delay time).
looptime | Filter looptime in seconds. |
revtime | Reverb time in seconds. |
- DaisySP
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This is the complete list of members for daisysp::Balance, including all inherited members.
-Balance() (defined in daisysp::Balance) | daisysp::Balance | inline |
Init(float sample_rate) | daisysp::Balance | |
Process(float sig, float comp) | daisysp::Balance | |
SetCutoff(float cutoff) | daisysp::Balance | inline |
~Balance() (defined in daisysp::Balance) | daisysp::Balance | inline |
- DaisySP
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#include <balance.h>
-Public Member Functions | |
void | Init (float sample_rate) |
float | Process (float sig, float comp) |
void | SetCutoff (float cutoff) |
Balances two sound sources. Sig is boosted to the level of comp.
-Original author(s) : Barry Vercoe, john ffitch, Gabriel Maldonado
-Year: 1991
-Ported from soundpipe by Ben Sergentanis, May 2020
-Initializes the balance module.
sample_rate | - The sample rate of the audio engine being run. |
adjust sig level to level of comp
- -adjusts the rate at which level compensation happens
cutoff | : Sets half power point of special internal cutoff filter. |
defaults to 10
- -
- DaisySP
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This is the complete list of members for daisysp::Biquad, including all inherited members.
-Biquad() (defined in daisysp::Biquad) | daisysp::Biquad | inline |
Init(float sample_rate) | daisysp::Biquad | |
Process(float in) | daisysp::Biquad | |
SetCutoff(float cutoff) | daisysp::Biquad | inline |
SetRes(float res) | daisysp::Biquad | inline |
~Biquad() (defined in daisysp::Biquad) | daisysp::Biquad | inline |
- DaisySP
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#include <biquad.h>
-Public Member Functions | |
void | Init (float sample_rate) |
float | Process (float in) |
void | SetRes (float res) |
void | SetCutoff (float cutoff) |
Two pole recursive filter
-Original author(s) : Hans Mikelson
-Year: 1998
-Ported from soundpipe by Ben Sergentanis, May 2020
-Initializes the biquad module.
sample_rate | - The sample rate of the audio engine being run. |
Filters the input signal
Sets filter cutoff in Hz
cutoff | : Set filter cutoff. |
Sets resonance amount
res | : Set filter resonance. |
- DaisySP
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This is the complete list of members for daisysp::Bitcrush, including all inherited members.
-Bitcrush() (defined in daisysp::Bitcrush) | daisysp::Bitcrush | inline |
Init(float sample_rate) | daisysp::Bitcrush | |
Process(float in) | daisysp::Bitcrush | |
SetBitDepth(int bitdepth) | daisysp::Bitcrush | inline |
SetCrushRate(float crushrate) | daisysp::Bitcrush | inline |
~Bitcrush() (defined in daisysp::Bitcrush) | daisysp::Bitcrush | inline |
- DaisySP
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#include <bitcrush.h>
-Public Member Functions | |
void | Init (float sample_rate) |
float | Process (float in) |
void | SetBitDepth (int bitdepth) |
void | SetCrushRate (float crushrate) |
bitcrush module
-Original author(s) : Paul Batchelor,
-Ported from soundpipe by Ben Sergentanis, May 2020
-Initializes the bitcrush module.
sample_rate | - The sample rate of the audio engine being run. |
bit crushes and downsamples the input
- -adjusts bitdepth
bitdepth | : Sets bit depth, 0...16 |
adjusts the downsampling frequency
crushrate | : Sets rate to downsample to, 0...SampleRate |
- DaisySP
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This is the complete list of members for daisysp::BlOsc, including all inherited members.
-BlOsc() (defined in daisysp::BlOsc) | daisysp::BlOsc | inline |
Init(float sample_rate) | daisysp::BlOsc | |
Process() | daisysp::BlOsc | |
Reset() | daisysp::BlOsc | |
SetAmp(float amp) | daisysp::BlOsc | inline |
SetFreq(float freq) | daisysp::BlOsc | inline |
SetPw(float pw) | daisysp::BlOsc | inline |
SetWaveform(uint8_t waveform) | daisysp::BlOsc | inline |
WAVE_OFF enum value (defined in daisysp::BlOsc) | daisysp::BlOsc | |
WAVE_SAW enum value (defined in daisysp::BlOsc) | daisysp::BlOsc | |
WAVE_SQUARE enum value (defined in daisysp::BlOsc) | daisysp::BlOsc | |
WAVE_TRIANGLE enum value (defined in daisysp::BlOsc) | daisysp::BlOsc | |
Waveforms enum name | daisysp::BlOsc | |
~BlOsc() (defined in daisysp::BlOsc) | daisysp::BlOsc | inline |
- DaisySP
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This is the complete list of members for daisysp::Comb, including all inherited members.
-Comb() (defined in daisysp::Comb) | daisysp::Comb | inline |
Init(float sample_rate, float *buff, size_t size) | daisysp::Comb | |
Process(float in) | daisysp::Comb | |
SetFreq(float freq) | daisysp::Comb | inline |
SetPeriod(float looptime) | daisysp::Comb | |
SetRevTime(float revtime) | daisysp::Comb | inline |
~Comb() (defined in daisysp::Comb) | daisysp::Comb | inline |
- DaisySP
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#include <comb.h>
-Public Member Functions | |
void | Init (float sample_rate, float *buff, size_t size) |
float | Process (float in) |
void | SetPeriod (float looptime) |
void | SetFreq (float freq) |
void | SetRevTime (float revtime) |
void Comb::Init | -( | -float | sample_rate, | -
- | - | float * | buff, | -
- | - | size_t | size ) | -
Initializes the Comb module.
sample_rate | - The sample rate of the audio engine being run. |
buff | - input buffer, kept in either main() or global space |
size | - size of buff |
Sets the frequency of the comb filter in Hz
- -Sets the period of the comb filter in seconds
- -Sets the decay time of the comb filter
- -
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This is the complete list of members for daisysp::Compressor, including all inherited members.
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#include <compressor.h>
-Public Member Functions | |
void | Init (float sample_rate) |
float | Process (float in) |
float | Process (float in, float key) |
float | Apply (float in) |
void | ProcessBlock (float *in, float *out, size_t size) |
void | ProcessBlock (float *in, float *out, float *key, size_t size) |
void | ProcessBlock (float **in, float **out, float *key, size_t channels, size_t size) |
float | GetRatio () |
void | SetRatio (float ratio) |
float | GetThreshold () |
void | SetThreshold (float threshold) |
float | GetAttack () |
void | SetAttack (float attack) |
float | GetRelease () |
void | SetRelease (float release) |
float | GetMakeup () |
void | SetMakeup (float gain) |
void | AutoMakeup (bool enable) |
float | GetGain () |
dynamics compressor
-influenced by compressor in soundpipe (from faust).
-Modifications made to do:
by: shensley, improved upon by AvAars
Apply compression to the audio signal, based on the previously calculated gain
in | audio input signal |
Enables or disables the automatic makeup gain. Disabling sets the makeup gain to 0.0
enable | true to enable, false to disable |
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Gets the envelope time for onset of compression
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Gets the gain reduction in dB
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Gets the additional gain to make up for the compression
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Gets the amount of gain reduction
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Gets the envelope time for release of compression
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Gets the threshold in dB
- -Initializes compressor
sample_rate | rate at which samples will be produced by the audio engine. |
Compress the audio input signal, saves the calculated gain
in | audio input signal |
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Compresses the audio input signal, keyed by a secondary input.
in | audio input signal (to be compressed) |
key | audio input that will be used to side-chain the compressor |
void Compressor::ProcessBlock | -( | -float ** | in, | -
- | - | float ** | out, | -
- | - | float * | key, | -
- | - | size_t | channels, | -
- | - | size_t | size ) | -
Compresses a block of multiple channels of audio, keyed by a secondary input
in | audio input signals (to be compressed) |
out | audio output signals |
key | audio input that will be used to side-chain the compressor |
channels | the number of audio channels |
size | the size of the block |
void Compressor::ProcessBlock | -( | -float * | in, | -
- | - | float * | out, | -
- | - | float * | key, | -
- | - | size_t | size ) | -
Compresses a block of audio, keyed by a secondary input
in | audio input signal (to be compressed) |
out | audio output signal |
key | audio input that will be used to side-chain the compressor |
size | the size of the block |
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Compresses a block of audio
in | audio input signal |
out | audio output signal |
size | the size of the block |
Sets the envelope time for onset of compression for signals above the threshold.
attack | Expects 0.001 -> 10 |
Manually sets the additional gain to make up for the compression
gain | Expects 0.0 -> 80 |
Sets the amount of gain reduction applied to compressed signals
ratio | Expects 1.0 -> 40. (untested with values < 1.0) |
Sets the envelope time for release of compression as input signal falls below threshold.
release | Expects 0.001 -> 10 |
Sets the threshold in dB at which compression will be applied
threshold | Expects 0.0 -> -80. |
- DaisySP
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This is the complete list of members for daisysp::Fold, including all inherited members.
-Fold() (defined in daisysp::Fold) | daisysp::Fold | inline |
Init() | daisysp::Fold | |
Process(float in) | daisysp::Fold | |
SetIncrement(float incr) | daisysp::Fold | inline |
~Fold() (defined in daisysp::Fold) | daisysp::Fold | inline |
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#include <fold.h>
-Public Member Functions | |
void | Init () |
float | Process (float in) |
void | SetIncrement (float incr) |
fold module
-Original author(s) : John FFitch, Gabriel Maldonado
-Year : 1998
-Ported from soundpipe by Ben Sergentanis, May 2020
-void Fold::Init | -( | -) | -- |
Initializes the fold module.
- -incr | : set fold increment |
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This is the complete list of members for daisysp::Jitter, including all inherited members.
-Init(float sample_rate) | daisysp::Jitter | |
Jitter() (defined in daisysp::Jitter) | daisysp::Jitter | inline |
Process() | daisysp::Jitter | |
SetAmp(float amp) | daisysp::Jitter | |
SetCpsMax(float cps_max) | daisysp::Jitter | |
SetCpsMin(float cps_min) | daisysp::Jitter | |
~Jitter() (defined in daisysp::Jitter) | daisysp::Jitter | inline |
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#include <jitter.h>
-Public Member Functions | |
void | Init (float sample_rate) |
float | Process () |
void | SetCpsMin (float cps_min) |
void | SetCpsMax (float cps_max) |
void | SetAmp (float amp) |
Randomly segmented line generator
- Originally extracted from csound by Paul Batchelor.
- Ported by Ben Sergentanis, June 2020
@year 1998
-Location: Opcodes/uggab.c (csound)
-Initializes Jitter module
sample_rate | Audio engine sample rate |
float Jitter::Process | -( | -) | -- |
Get next floating point jitter sample
- -Set the amplitude of the jitter. Jitters fall from -amp to +amp
amp | Jitter amplitude |
Set the maximum speed of the jitter engine.
cps_max | Maximum number of jitters per second. |
Set the minimum speed of the jitter engine.
cps_min | Number of new jitters per second |
- DaisySP
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This is the complete list of members for daisysp::Line, including all inherited members.
-Init(float sample_rate) | daisysp::Line | |
Line() (defined in daisysp::Line) | daisysp::Line | inline |
Process(uint8_t *finished) | daisysp::Line | |
Start(float start, float end, float dur) | daisysp::Line | |
~Line() (defined in daisysp::Line) | daisysp::Line | inline |
- DaisySP
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#include <line.h>
-Public Member Functions | |
void | Init (float sample_rate) |
float | Process (uint8_t *finished) |
void | Start (float start, float end, float dur) |
creates a Line segment signal
-Normal Mode: Input is added to the existing loop infinitely while recording
-Onetime Dub Mode: Recording starts at the first sample of the buffer and is added to the existing buffer contents. Recording automatically stops after one full loop.
-Replace Mode: Audio in the buffer is replaced while recording is on.
-Frippertronics Mode: infinite looping recording with fixed decay on each loop. The module acts like tape-delay set up.
+Normal Mode: Input is added to the existing loop infinitely while recording
+Onetime Dub Mode: Recording starts at the first sample of the buffer and is added to the existing buffer contents. Recording automatically stops after one full loop.
+Replace Mode: Audio in the buffer is replaced while recording is on.
+Frippertronics Mode: infinite looping recording with fixed decay on each loop. The module acts like tape-delay set up.
Increments the Mode by one step useful for buttons, etc. that need to step through the Looper modes.
+Increments the Mode by one step useful for buttons, etc. that need to step through the Looper modes.
Sets the recording mode to the specified Mode.
+Sets the recording mode to the specified Mode.
@@ -355,7 +355,7 @@Engages/Disengages the recording, depending on Mode. In all modes, the first time this is triggered a new loop will be started. The second trigger will set the loop size, and begin playback of the loop.
+Engages/Disengages the recording, depending on Mode. In all modes, the first time this is triggered a new loop will be started. The second trigger will set the loop size, and begin playback of the loop.
diff --git a/classdaisysp_1_1_mode-members.html b/classdaisysp_1_1_mode-members.html deleted file mode 100644 index d88d0943..00000000 --- a/classdaisysp_1_1_mode-members.html +++ /dev/null @@ -1,97 +0,0 @@ - - - - - - - -
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This is the complete list of members for daisysp::Mode, including all inherited members.
-Clear() | daisysp::Mode | |
Init(float sample_rate) | daisysp::Mode | |
Mode() (defined in daisysp::Mode) | daisysp::Mode | inline |
Process(float in) | daisysp::Mode | |
SetFreq(float freq) | daisysp::Mode | inline |
SetQ(float q) | daisysp::Mode | inline |
~Mode() (defined in daisysp::Mode) | daisysp::Mode | inline |
- DaisySP
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#include <mode.h>
-Public Member Functions | |
void | Init (float sample_rate) |
float | Process (float in) |
void | Clear () |
void | SetFreq (float freq) |
void | SetQ (float q) |
Resonant Modal Filter
-Extracted from soundpipe to work as a Daisy Module,
-originally extracted from csound by Paul Batchelor.
-Original Author(s): Francois Blanc, Steven Yi
-Year: 2001
-Location: Opcodes/biquad.c (csound)
-void Mode::Clear | -( | -) | -- |
Clears the filter, returning the output to 0.0
- -Initializes the instance of the module. sample_rate: frequency of the audio engine in Hz
- -Processes one input sample through the filter, and returns the output.
- -Sets the resonant frequency of the modal filter. Range: Any frequency such that sample_rate / freq < PI (about 15.2kHz at 48kHz)
- -Sets the quality factor of the filter. Range: Positive Numbers (Good values range from 70 to 1400)
- -
- DaisySP
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This is the complete list of members for daisysp::MoogLadder, including all inherited members.
-Init(float sample_rate) | daisysp::MoogLadder | |
MoogLadder() (defined in daisysp::MoogLadder) | daisysp::MoogLadder | inline |
Process(float in) | daisysp::MoogLadder | |
SetFreq(float freq) | daisysp::MoogLadder | inline |
SetRes(float res) | daisysp::MoogLadder | inline |
~MoogLadder() (defined in daisysp::MoogLadder) | daisysp::MoogLadder | inline |
- DaisySP
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#include <moogladder.h>
-Public Member Functions | |
void | Init (float sample_rate) |
float | Process (float in) |
void | SetFreq (float freq) |
void | SetRes (float res) |
Moog ladder filter module
-Ported from soundpipe
-Original author(s) : Victor Lazzarini, John ffitch (fast tanh), Bob Moog
-Initializes the MoogLadder module. sample_rate - The sample rate of the audio engine being run.
- -Sets the cutoff frequency or half-way point of the filter. Arguments
Sets the resonance of the filter.
- -
- DaisySP
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#include <nlfilt.h>
-Public Member Functions | |
void | Init () |
void | ProcessBlock (float *in, float *out, size_t size) |
void | SetCoefficients (float a, float b, float d, float C, float L) |
void | SetA (float a) |
void | SetB (float b) |
void | SetD (float d) |
void | SetC (float C) |
void | SetL (float L) |
Non-linear filter
-port by: Stephen Hensley, December 2019
-The four 5-coefficients: a, b, d, C, and L are used to configure different filter types.
-Structure for Dobson/Fitch nonlinear filter
-Revised Formula from Risto Holopainen 12 Mar 2004
-Y{n} =tanh(a Y{n-1} + b Y{n-2} + d Y^2{n-L} + X{n} - C)
Though traditional filter types can be made, the effect will always respond differently to different input.
-This Source is a heavily modified version of the original source from Csound.
-void NlFilt::ProcessBlock | -( | -float * | in, | -
- | - | float * | out, | -
- | - | size_t | size ) | -
Process the array pointed to by *in and updates the output to *out; This works on a block of audio at once, the size of which is set with the size.
- -Set Coefficient a
- -Set Coefficient b
- -Set Coefficient C
- -
-
|
- -inline | -
inputs these are the five coefficients for the filter.
- -Set Coefficient d
- -Set Coefficient L
- -This is the complete list of members for daisysp::NlFilt, including all inherited members.
+This is the complete list of members for daisysp::OnePole, including all inherited members.
Init() | daisysp::NlFilt | |
ProcessBlock(float *in, float *out, size_t size) | daisysp::NlFilt | |
SetA(float a) | daisysp::NlFilt | inline |
SetB(float b) | daisysp::NlFilt | inline |
SetC(float C) | daisysp::NlFilt | inline |
SetCoefficients(float a, float b, float d, float C, float L) | daisysp::NlFilt | inline |
SetD(float d) | daisysp::NlFilt | inline |
SetL(float L) | daisysp::NlFilt | inline |
FILTER_MODE_HIGH_PASS enum value (defined in daisysp::OnePole) | daisysp::OnePole | |
FILTER_MODE_LOW_PASS enum value (defined in daisysp::OnePole) | daisysp::OnePole | |
FilterMode enum name | daisysp::OnePole | |
Init() | daisysp::OnePole | inline |
OnePole() (defined in daisysp::OnePole) | daisysp::OnePole | inline |
Process(float in) | daisysp::OnePole | inline |
ProcessBlock(float *in_out, size_t size) | daisysp::OnePole | inline |
Reset() | daisysp::OnePole | inline |
SetFilterMode(FilterMode mode) | daisysp::OnePole | inline |
SetFrequency(float freq) | daisysp::OnePole | inline |
~OnePole() (defined in daisysp::OnePole) | daisysp::OnePole | inline |
#include <blosc.h>
One Pole Lowpass / Highpass Filter. + More...
+ +#include <onepole.h>
Public Types | |
enum | Waveforms { WAVE_TRIANGLE -, WAVE_SAW -, WAVE_SQUARE -, WAVE_OFF + |
enum | FilterMode { FILTER_MODE_LOW_PASS +, FILTER_MODE_HIGH_PASS } |
Operational modes of the filter. More... | |
Public Member Functions | |
void | Init (float sample_rate) |
float | Process () |
void | SetFreq (float freq) |
void | SetAmp (float amp) |
void | SetPw (float pw) |
void | SetWaveform (uint8_t waveform) |
void | Reset () |
void | Init () |
void | Reset () |
void | SetFrequency (float freq) |
void | SetFilterMode (FilterMode mode) |
float | Process (float in) |
void | ProcessBlock (float *in_out, size_t size) |
Band Limited Oscillator
-Based on bltriangle, blsaw, blsquare from soundpipe
-Original Author(s): Paul Batchelor, saw2 Faust by Julius Smith
-Ported by Ben Sergentanis, May 2020
+One Pole Lowpass / Highpass Filter.
+ +Bl Waveforms
--Initialize oscillator. -Defaults to: 440Hz, .5 amplitude, .5 pw, Triangle.
+Operational modes of the filter.
+Initializes the module
Process audio through the filter
in | The next sample to be processed |
void daisysp::BlOsc::SetAmp | +void daisysp::OnePole::ProcessBlock | ( | -float | amp | ) | +float * | in_out, | +
+ | size_t | size ) |
Process a block of audio through the filter
in_out | Pointer to the block of samples to be processed |
size | Size of the block of samples to be processed. |
void daisysp::BlOsc::SetFreq | +void daisysp::OnePole::Reset | ( | -float | freq | ) | +) |
Reset the module to its default state
void daisysp::BlOsc::SetPw | +void daisysp::OnePole::SetFilterMode | ( | -float | pw | ) | +FilterMode | mode | ) |
Set the filter mode
mode | Filter mode. Can be lowpass or highpass |
void daisysp::BlOsc::SetWaveform | +void daisysp::OnePole::SetFrequency | ( | -uint8_t | waveform | ) | +float | freq | ) |
Set the filter cutoff frequency
freq | Cutoff frequency. Valid range from 0 to .497f |
Public Types | |
enum | { + |
enum | { WAVE_SIN , WAVE_TRI , WAVE_SAW @@ -101,7 +101,7 @@ WAVE_LAST } |
Public Member Functions |
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This is the complete list of members for daisysp::Pluck, including all inherited members.
-GetAmp() | daisysp::Pluck | inline |
GetDamp() | daisysp::Pluck | inline |
GetDecay() | daisysp::Pluck | inline |
GetFreq() | daisysp::Pluck | inline |
GetMode() | daisysp::Pluck | inline |
Init(float sample_rate, float *buf, int32_t npt, int32_t mode) | daisysp::Pluck | |
Pluck() (defined in daisysp::Pluck) | daisysp::Pluck | inline |
Process(float &trig) | daisysp::Pluck | |
SetAmp(float amp) | daisysp::Pluck | inline |
SetDamp(float damp) | daisysp::Pluck | inline |
SetDecay(float decay) | daisysp::Pluck | inline |
SetFreq(float freq) | daisysp::Pluck | inline |
SetMode(int32_t mode) | daisysp::Pluck | inline |
~Pluck() (defined in daisysp::Pluck) | daisysp::Pluck | inline |
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#include <pluck.h>
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void | Init (float sample_rate, float *buf, int32_t npt, int32_t mode) |
float | Process (float &trig) |
void | SetAmp (float amp) |
void | SetFreq (float freq) |
void | SetDecay (float decay) |
void | SetDamp (float damp) |
void | SetMode (int32_t mode) |
float | GetAmp () |
float | GetFreq () |
float | GetDecay () |
float | GetDamp () |
int32_t | GetMode () |
Produces a naturally decaying plucked string or drum sound based on the Karplus-Strong algorithms.
-Ported from soundpipe to DaisySP
-This code was originally extracted from the Csound opcode "pluck"
-Original Author(s): Barry Vercoe, John ffitch Year: 1991
-Location: OOps/ugens4.c
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Returns the current value for amp.
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Returns the current value for damp.
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Returns the current value for decay.
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Returns the current value for freq.
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Returns the current value for mode.
- -void Pluck::Init | -( | -float | sample_rate, | -
- | - | float * | buf, | -
- | - | int32_t | npt, | -
- | - | int32_t | mode ) | -
Initializes the Pluck module.
\param sample_rate: Sample rate of the audio engine being run. -\param buf: buffer used as an impulse when triggering the Pluck algorithm -\param npt: number of elementes in buf. -\param mode: Sets the mode of the algorithm. --
Processes the waveform to be generated, returning one sample. This should be called once per sample period.
- -Sets the amplitude of the output signal. Input range: 0-1?
- -Sets the dampening factor applied by the filter (based on PLUCK_MODE) Input range: 0-1
- -Sets the time it takes for a triggered note to end in seconds. Input range: 0-1
- -Sets the frequency of the output signal in Hz. Input range: Any positive value
- -Sets the mode of the algorithm.
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This is the complete list of members for daisysp::PolyPluck< num_voices >, including all inherited members.
-Init(float sample_rate) | daisysp::PolyPluck< num_voices > | inline |
Process(float &trig, float note) | daisysp::PolyPluck< num_voices > | inline |
SetDecay(float p) | daisysp::PolyPluck< num_voices > | inline |
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#include <PolyPluck.h>
-Public Member Functions | |
void | Init (float sample_rate) |
float | Process (float &trig, float note) |
void | SetDecay (float p) |
Simplified Pseudo-Polyphonic Pluck Voice
-Template Based Pluck Voice, with configurable number of voices and simple pseudo-polyphony.
-DC Blocking included to prevent biases from causing unwanted saturation distortion.
-Author**: shensley
-Date Added**: March 2020
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Process function, synthesizes and sums the output of all voices, triggering a new voice with frequency of MIDI note number when trig > 0.
-trig | value by reference of trig. When trig > 0 a the next voice will be triggered, and trig will be set to 0. |
note | MIDI note number for the active_voice. |
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Sets the decay coefficients of the pluck voices.
p | expects 0.0-1.0 input. |
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This is the complete list of members for daisysp::Port, including all inherited members.
-GetHtime() | daisysp::Port | inline |
Init(float sample_rate, float htime) | daisysp::Port | |
Port() (defined in daisysp::Port) | daisysp::Port | inline |
Process(float in) | daisysp::Port | |
SetHtime(float htime) | daisysp::Port | inline |
~Port() (defined in daisysp::Port) | daisysp::Port | inline |
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#include <port.h>
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void | Init (float sample_rate, float htime) |
float | Process (float in) |
void | SetHtime (float htime) |
float | GetHtime () |
Applies portamento to an input signal.
-At each new step value, the input is low-pass filtered to move towards that value at a rate determined by ihtim. ihtim is the half-time of the function (in seconds), during which the curve will traverse half the distance towards the new value, then half as much again, etc., theoretically never reaching its asymptote.
-This code has been ported from Soundpipe to DaisySP by Paul Batchelor.
-The Soundpipe module was extracted from the Csound opcode "portk".
-Original Author(s): Robbin Whittle, John ffitch
-Year: 1995, 1998
-Location: Opcodes/biquad.c
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returns current value of htime
- -Initializes Port module
-sample_rate | sample rate of audio engine |
htime | half-time of the function, in seconds. |
Applies portamento to input signal and returns processed signal.
Sets htime
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This is the complete list of members for daisysp::ReverbSc, including all inherited members.
-Init(float sample_rate) | daisysp::ReverbSc | |
Process(const float &in1, const float &in2, float *out1, float *out2) | daisysp::ReverbSc | |
ReverbSc() (defined in daisysp::ReverbSc) | daisysp::ReverbSc | inline |
SetFeedback(const float &fb) | daisysp::ReverbSc | inline |
SetLpFreq(const float &freq) | daisysp::ReverbSc | inline |
~ReverbSc() (defined in daisysp::ReverbSc) | daisysp::ReverbSc | inline |
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#include <reverbsc.h>
-Public Member Functions | |
int | Init (float sample_rate) |
int | Process (const float &in1, const float &in2, float *out1, float *out2) |
void | SetFeedback (const float &fb) |
void | SetLpFreq (const float &freq) |
Stereo Reverb
-Reverb SC: Ported from csound/soundpipe
-Original author(s): Sean Costello, Istvan Varga
-Year: 1999, 2005
-Ported to soundpipe by: Paul Batchelor
-Ported by: Stephen Hensley
-Initializes the reverb module, and sets the sample_rate at which the Process function will be called. Returns 0 if all good, or 1 if it runs out of delay times exceed maximum allowed.
- -int ReverbSc::Process | -( | -const float & | in1, | -
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Process the input through the reverb, and updates values of out1, and out2 with the new processed signal.
- -controls the reverb time. reverb tail becomes infinite when set to 1.0
fb | - sets reverb time. range: 0.0 to 1.0 |
controls the internal dampening filter's cutoff frequency.
freq | - low pass frequency. range: 0.0 to sample_rate / 2 |
Comb filter / KS string. +
Comb filter / KS string. More...
#include <KarplusString.h>
Comb filter / KS string.
+Comb filter / KS string.
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This is the complete list of members for daisysp::Tone, including all inherited members.
-GetFreq() | daisysp::Tone | inline |
Init(float sample_rate) | daisysp::Tone | |
Process(float in) | daisysp::Tone | |
SetFreq(float freq) | daisysp::Tone | inline |
Tone() (defined in daisysp::Tone) | daisysp::Tone | inline |
~Tone() (defined in daisysp::Tone) | daisysp::Tone | inline |
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#include <tone.h>
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void | Init (float sample_rate) |
float | Process (float in) |
void | SetFreq (float freq) |
float | GetFreq () |
A first-order recursive low-pass filter with variable frequency response.
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Initializes the Tone module. sample_rate - The sample rate of the audio engine being run.
- -Processes one sample through the filter and returns one sample. in - input signal
- -Sets the cutoff frequency or half-way point of the filter.
-freq | - frequency value in Hz. Range: Any positive value. |
mode | Mode to set. Works best -1 to 1 |
mode | Mode to set. Works best -1 to 1 |
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blosc.h | |
fm2.h | |
formantosc.h |
Files | |
balance.h | |
compressor.h | |
crossfade.h | |
limiter.h | |
dsp.h | |
jitter.h | |
looper.h | |
maytrig.h | |
metro.h | |
port.h | |
samplehold.h | |
smooth_random.h |
Files | |
allpass.h | |
atone.h | |
biquad.h | |
comb.h | |
fir.h | |
mode.h | |
moogladder.h | |
nlfilt.h | |
onepole.h | |
soap.h | |
svf.h | |
tone.h | |
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The method of natural decay that the algorithm will use.
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This is the complete list of members for daisysp::ReverbScDl, including all inherited members.
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#include <reverbsc.h>
-Public Attributes | |
int | write_pos |
int | buffer_size |
int | read_pos |
int | read_pos_frac |
int | read_pos_frac_inc |
int | dummy |
int | seed_val |
int | rand_line_cnt |
float | filter_state |
float * | buf |
Delay line for internal reverb use
-float* daisysp::ReverbScDl::buf | -
buffer ptr
- -int daisysp::ReverbScDl::buffer_size | -
buffer size
- -int daisysp::ReverbScDl::dummy | -
dummy var
- -float daisysp::ReverbScDl::filter_state | -
state of filter
- -int daisysp::ReverbScDl::rand_line_cnt | -
number of random lines
- -int daisysp::ReverbScDl::read_pos | -
read position
- -int daisysp::ReverbScDl::read_pos_frac | -
fractional component of read pos
- -int daisysp::ReverbScDl::read_pos_frac_inc | -
increment for fractional
- -int daisysp::ReverbScDl::seed_val | -
randseed
- -int daisysp::ReverbScDl::write_pos | -
write position
- -- Maybe make this an ADsr_ that has AD/AR/Asr_ modes.
-Selecting which channels should be initialized/included in the sequence conversion.
Setup a similar start function for an external mux, but that seems outside the scope of this file.
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